80gvp_us

Genesys Voice Platform 8.0

User’s Guide

The information contained herein is proprietary and confidential and cannot be disclosed or duplicated without the prior written consent of Genesys Telecommunications Laboratories, Inc.Copyright © 2008 Genesys Telecommunications Laboratories, Inc. All rights reserved.

About GenesysGenesys Telecommunications Laboratories, Inc., a subsidiary of Alcatel-Lucent, is 100% focused on software for call centers. Genesys recognizes that better interactions drive better business and build company reputations. Customer service solutions from Genesys deliver on this promise for Global 2000 enterprises, government organizations, and telecommunications service providers across 80 countries, directing more than 100 million customer interactions every day. Sophisticated routing and reporting across voice, e-mail, and Web channels ensure that customers are quickly connected to the best available resource—the first time. Genesys offers solutions for customer service, help desks, order desks, collections, outbound telesales and service, and workforce management. Visit www.genesyslab.com for more information.Each product has its own documentation for online viewing at the Genesys Technical Support website or on the Documentation Library DVD, which is available from Genesys upon request. For more information, contact your sales representative.

NoticeAlthough reasonable effort is made to ensure that the information in this document is complete and accurate at the time of release, Genesys Telecommunications Laboratories, Inc., cannot assume responsibility for any existing errors. Changes and/or corrections to the information contained in this document may be incorporated in future versions.

Your Responsibility for Your System’s SecurityYou are responsible for the security of your system. Product administration to prevent unauthorized use is your responsibility. Your system administrator should read all documents provided with this product to fully understand the features available that reduce your risk of incurring charges for unlicensed use of Genesys products.

TrademarksGenesys, the Genesys logo, and T-Server are registered trademarks of Genesys Telecommunications Laboratories, Inc. All other trademarks and trade names referred to in this document are the property of other companies. The Crystal monospace font is used by permission of Software Renovation Corporation, www.SoftwareRenovation.com.

Technical Support from VARsIf you have purchased support from a value-added reseller (VAR), please contact the VAR for technical support.

Technical Support from GenesysIf you have purchased support directly from Genesys, please contact Genesys Technical Support at the following regional numbers:

Prior to contacting technical support, please refer to the Genesys Technical Support Guide for complete contact information and procedures.

Ordering and Licensing InformationComplete information on ordering and licensing Genesys products can be found in the Genesys 7 Licensing Guide.

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User’s Guide 3

Table of ContentsList of Procedures................................................................................... 9

Preface ................................................................................................................. 11Intended Audience................................................................................... 12Chapter Summaries................................................................................. 12Document Conventions ........................................................................... 13Related Resources .................................................................................. 15Making Comments on This Document .................................................... 17

Part 1 Overview ...................................................................... 19

Chapter 1 Introduction........................................................................................... 21

About GVP............................................................................................... 21GVP Components............................................................................... 21IVR Profiles......................................................................................... 23GVP MIBs ........................................................................................... 23

Genesys Administrator ............................................................................ 23

Chapter 2 How GVP Works ................................................................................... 25

Sample Call Flow..................................................................................... 25How the Resource Manager Works......................................................... 27

Session Management ......................................................................... 27Service Selection ................................................................................ 29Policy Enforcement............................................................................. 32Service Request Modification ............................................................. 32Resource Management ...................................................................... 32High Availability and Scalability .......................................................... 37

How the Media Control Platform Works .................................................. 37About the Media Control Platform....................................................... 38Operational Overview ......................................................................... 38Media Services ................................................................................... 42Speech Services ................................................................................. 44Transfers............................................................................................. 45

Table of Contents

4 Genesys Voice Platform 8.0

Conferencing ...................................................................................... 50Debugging VoiceXML Applications..................................................... 50

How the Call Control Platform Works ...................................................... 50About the Call Control Platform .......................................................... 50Operational Overview ......................................................................... 51Device Profiles.................................................................................... 53

How the Fetching Module Works............................................................. 55About the Fetching Module................................................................. 55Caching............................................................................................... 56

Logging and Reporting ............................................................................ 60EMS Reporting Architecture ............................................................... 62EMS Logging ...................................................................................... 63CDR Reporting ................................................................................... 66OR Service ......................................................................................... 68Reporting Client .................................................................................. 68Reporting Server................................................................................. 68Reporting Web Services ..................................................................... 69

SNMP Monitoring .................................................................................... 70Secure Communications ......................................................................... 70

Considerations and Usage Notes ....................................................... 71GVP Identifiers and SIP Headers ............................................................ 71

Session Identifiers .............................................................................. 72Application Identifiers.......................................................................... 72

Part 2 Provisioning GVP........................................................ 75

Chapter 3 Configuration and Provisioning Overview......................................... 77

Configuring GVP...................................................................................... 77Configuring GVP Processes in the Genesys Administrator ................ 78

Task Summary: Configuring GVP............................................................ 81

Chapter 4 Configuring Common Features........................................................... 85

Configuring SIP Communications and Routing ....................................... 86Enabling Secure Communications .......................................................... 93Enabling Conference Services .............................................................. 102Configuring EMS Reporting................................................................... 103Configuring Logging ...............................................................................111Customizing SIP Responses ................................................................. 116Configuring Session Timers and Timeouts ............................................ 117

Resource Manager Session Timers.................................................. 118Additional Timeouts .......................................................................... 120

User’s Guide 5

Table of Contents

Chapter 5 Configuring the Resource Manager.................................................. 123

Task Summary: Configuring the Resource Manager............................. 123Important Resource Manager Configuration Options ............................ 124Configuring Logical Resource Groups................................................... 126Enabling High Availability ...................................................................... 134

Chapter 6 Provisioning IVR Profiles................................................................... 139

Provisioning IVR Profiles for GVP ......................................................... 139IVR Profile Configuration Options.......................................................... 141

gvp.general Section .......................................................................... 141gvp.log Section ................................................................................. 142gvp.policy Section............................................................................. 143gvp.policy.dialing-rules...................................................................... 148gvp.service-parameters Section ....................................................... 149gvp.service-prerequisite Section....................................................... 150

Mapping IVR Profiles to Dialed Numbers .............................................. 151Specifying Tenant Environment Settings ............................................... 153

gvp.dnis-range Section ..................................................................... 154gvp.general Section .......................................................................... 154gvp.policy Section............................................................................. 155gvp.service-parameters Section ....................................................... 155

Chapter 7 Configuring the Media Control Platform .......................................... 157

Task Summary: Configuring the Media Control Platform....................... 157Enabling ASR and TTS ......................................................................... 159Important Media Control Platform Configuration Options...................... 162Important MRCP Server Configuration Options .................................... 178

Chapter 8 Configuring the Call Control Platform.............................................. 183

Task Summary: Configuring the Call Control Platform .......................... 183Important Call Control Platform Configuration Options.......................... 185Configuring Device Profiles ................................................................... 188

Device Profile Configuration File ...................................................... 189Customizing Device Profiles ............................................................. 193

Chapter 9 Configuring the Fetching Module and Squid Proxy........................ 197

Task Summary: Configuring the Fetching Module and Squid................ 197Important Fetching Module Configuration Options ................................ 198Configuring the Squid Caching Proxy.................................................... 200

Table of Contents


Chapter 10 Configuring the Reporting Server..................................................... 203

Task Summary: Configuring the Reporting Server ................................ 203Configuring Reporting, by Granularity ................................................... 204Configuring Database Retention Policies .............................................. 206Important Reporting Server Configuration Options ............................... 207Controlling Access to Reporting Services ............................................. 210Connecting to the Genesys Administrator ............................................. 213

Part 3 Monitoring GVP ......................................................... 217

Chapter 11 Reporting Overview............................................................................ 219

Genesys Administrator .......................................................................... 219Running a Report .................................................................................. 219Report Filters ......................................................................................... 224

Chapter 12 Real-Time Reports.............................................................................. 227

Overview................................................................................................ 227Active Call List ....................................................................................... 228

Chapter 13 Historical Reports............................................................................... 233

Overview................................................................................................ 233Historical Call Summary ........................................................................ 235Historical Peaks..................................................................................... 236Historical Call Browser .......................................................................... 238

Chapter 14 Voice Application Reports................................................................. 243

Overview................................................................................................ 243VAR Call Browser .................................................................................. 244Call Completion Summary..................................................................... 247IVR Action Usage .................................................................................. 248Last IVR Action Used ............................................................................ 249

Part 4 Appendixes................................................................ 253

Appendix A Module and Specifier IDs................................................................... 255

Media Control Platform.......................................................................... 255Next Generation Interpreter Module ID and Specifiers ..................... 265

User’s Guide 7

Table of Contents

Call Control Platform ............................................................................. 267Connection, Dialog, or Conference Events....................................... 267Media Controller Events.................................................................... 269Log_4 (INFO) Events ........................................................................ 271

Resource Manager ................................................................................ 271Fetching Module .................................................................................... 276

Appendix B SIP Response Codes.......................................................................... 279

SIP Responses within GVP................................................................... 279

Appendix C Media Control Platform Reference Information............................... 289

Audio and Video File Formats ............................................................... 289Audio-Only Formats—Play ............................................................... 289Video-Only Formats—Play ............................................................... 291Combined Audio and Video Formats—Play ..................................... 291Audio-Only Formats—Record........................................................... 292Video-Only Formats—Record........................................................... 293Combined Audio and Video Formats—Record................................. 294

Dynamic Media Control Platform Parameters ....................................... 294SIP Headers .......................................................................................... 295Handling Error Responses for Outbound Calls ..................................... 298VAR Metrics........................................................................................... 300

Appendix D Default Device Profiles....................................................................... 304

Appendix E Specifications and Standards ........................................................... 307

Specifications......................................................................................... 307Related Standards ................................................................................. 308Burke Draft Support............................................................................... 309

Appendix F Caching Reference Information ........................................................ 315

Caching Algorithms ............................................................................... 315Fetching Module Caching Algorithm................................................. 315Squid Caching Algorithm .................................................................. 316Squid Expiry Time Algorithm............................................................. 316

Squid Access Logs ................................................................................ 317

Index ............................................................................................................... 321

Table of Contents


User’s Guide 9

List of ProceduresViewing or modifying GVP configuration parameters . . . . . . . . . . . . . . . 78Creating an SSL private key and certificate . . . . . . . . . . . . . . . . . . . . . . 96Creating an SSL key and self-signed certificate for use with IIS . . . . . . 97Configuring the Fetching Module for HTTPS . . . . . . . . . . . . . . . . . . . . . 99Configuring a stand-alone Tomcat web server for SSL . . . . . . . . . . . . 101Configuring logical resource groups . . . . . . . . . . . . . . . . . . . . . . . . . . . 127Configuring Resource Manager High Availability . . . . . . . . . . . . . . . . . 134Setting up the cluster mode execution environment. . . . . . . . . . . . . . . 136Provisioning IVR Profiles . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 139Mapping IVR Profiles to DNs . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 152Provisioning ASR and TTS resources . . . . . . . . . . . . . . . . . . . . . . . . . 160Provisioning Device Profiles for the Call Control Platform . . . . . . . . . . 194Modifying the Squid Configuration . . . . . . . . . . . . . . . . . . . . . . . . . . . . 200Enabling HTTP Basic Authorization for Reporting . . . . . . . . . . . . . . . . 211Configuring the connection between the Reporting Server and the

Genesys Administrator . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 213Running a Report . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 219Running a Real-Time Report . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 227Running a Historical Report . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . . 233Running a Voice Application Report. . . . . . . . . . . . . . . . . . . . . . . . . . . 243

List of Procedures


User’s Guide 11

PrefaceWelcome to the Genesys Voice Platform 8.0 User’s Guide. This document provides detailed information about the configuring, provisioning, and monitoring of Genesys Voice Platform (GVP) and its components.This document is valid only for the 8.0 release.

Note: For releases of this document created for other releases of this product, please visit the Genesys Technical Support website, or request the Documentation Library DVD, which you can order by e-mail from Genesys Order Management at [email protected].

This preface provides an overview of this document, identifies the primary audience, introduces document conventions, and lists related reference information: It contains the following sections:

Intended Audience, page 12Chapter Summaries, page 12Document Conventions, page 13Related Resources, page 15Making Comments on This Document, page 17

GVP is a group of software components that constitute a robust, carrier-grade voice processing platform. GVP unifies voice and web technologies to provide a complete solution for customer self-service or assisted service.In the Voice Platform Solution (VPS), GVP 8.0 is fully integrated with the Genesys Management Framework. GVP uses the Genesys Administrator, the standard Genesys configuration and management Graphical User Interface (GUI), to configure, tune, activate, and manage GVP processes and GVP voice and call control applications. GVP interacts with other Genesys components and can be deployed in conjunction with other solutions, such as Enterprise Routing Solution (ERS), Network Routing Solution (NRS), and Network-based Contact Solution (NbCS).

Note: GVP is a scalable solution with flexible configuration and deployment options, but the initial release of GVP 8.0 is available for single-tenant configurations only.



Preface Intended Audience

Intended AudienceThis document, primarily intended for system integrators and administrators, assumes that you have a basic understanding of:• Computer-telephony integration (CTI) concepts, processes, terminology,

and applications.• Network design and operation.• Your own network configurations.You should also be familiar with the Genesys Framework architecture.

Chapter SummariesIn addition to this preface, this Deployment Guide contains the following chapters and appendixes:• Part 1: “Overview” on page 19

Chapter 1, “Introduction,” on page 21, provides a high-level description of the GVP architecture and the Genesys Administrator.Chapter 2, “How GVP Works,” on page 25, describes in detail how the GVP components perform their functions.

• Part 2: “Provisioning GVP” on page 75Chapter 3, “Configuration and Provisioning Overview,” on page 77, provides a general description of the steps to configure GVP components, as well as an overall summary of the tasks to configure GVP.Chapter 4, “Configuring Common Features,” on page 85, describes how to configure features that are configured across components.Chapter 5, “Configuring the Resource Manager,” on page 123, describes how to configure the Resource Manager, including configuring logical resource groups and High Availability.Chapter 6, “Provisioning IVR Profiles,” on page 139, describes how to configure the IVR Profiles and provision the VoiceXML and CCXML applications for the deployment.Chapter 7, “Configuring the Media Control Platform,” on page 157, describes how to configure the Media Control Platform, including enabling Automatic Speech Recognition (ASR) and Text-to-Speech (TTS).Chapter 8, “Configuring the Call Control Platform,” on page 183, describes how to configure the Call Control Platform and provision the device profiles.Chapter 9, “Configuring the Fetching Module and Squid Proxy,” on page 197, describes how to configure the Fetching Module and Squid Caching Proxy.

User’s Guide 13

Preface Document Conventions

Chapter 10, “Configuring the Reporting Server,” on page 203, describes how to configure the Reporting Server for important reporting behavior, such as granularity of reports and data retention policies.

• Part 3: “Monitoring GVP” on page 217Chapter 11, “Reporting Overview,” on page 219, describes how to run reports and filter report results.Chapter 12, “Real-Time Reports,” on page 227, describes the content of the real-time reports that are available in GVP.Chapter 13, “Historical Reports,” on page 233, describes the content of the historical reports that are available in GVP.Chapter 14, “Voice Application Reports,” on page 243, describes the Voice Application Reporter (VAR) reports that are available in GVP.

• Part 4: “Appendixes” on page 253Appendix A, “Module and Specifier IDs,” on page 255, provides module, specifier, and metrics ID information that is required for configuring certain reporting and logging settings.Appendix B, “SIP Response Codes,” on page 279 describes the SIP response codes that are generated by GVP components.Appendix C, “Media Control Platform Reference Information,” on page 289 provides detailed reference information that relates to Media Control Platform functioning.Appendix D, “Default Device Profiles,” on page 304 lists the properties of the device profiles that are provisioned by default.Appendix E, “Specifications and Standards,” on page 307 lists the W3C Voice Browser Working Group and Internet Engineering Task Force (IETF) standards that GVP supports.Appendix F, “Caching Reference Information,” on page 315 describes the caching algorithms that the Fetching Module and Squid caching proxy use, as well as the entries in the Squid access logs.

Document ConventionsThis document uses certain stylistic and typographical conventions—introduced here—that serve as shorthands for particular kinds of information.

Document Version Number

A version number appears at the bottom of the inside front cover of this document. Version numbers change as new information is added to this document. Here is a sample version number:

72fr_ref_09-2005_v7.2.000.00


Preface Document Conventions

You will need this number when you are talking with Genesys Technical Support about this product.

Type Styles

Italic

In this document, italic is used for emphasis, for documents’ titles, for definitions of (or first references to) unfamiliar terms, and for mathematical variables.

Examples: • Please consult the Genesys 7 Migration Guide for more information.• A customary and usual practice is one that is widely accepted and used

within a particular industry or profession.• Do not use this value for this option.• The formula, x +1 = 7 where x stands for . . .

Monospace Font

A monospace font, which looks like teletype or typewriter text, is used for all programming identifiers and GUI elements. This convention includes the names of directories, files, folders, configuration objects, paths, scripts, dialog boxes, options, fields, text and list boxes, operational modes, all buttons (including radio buttons), check boxes, commands, tabs, CTI events, and error messages; the values of options; logical arguments and command syntax; and code samples.

Examples: • Select the Show variables on screen check box.• Click the Summation button.• In the Properties dialog box, enter the value for the host server in your

environment.• In the Operand text box, enter your formula.• Click OK to exit the Properties dialog box.• The following table presents the complete set of error messages T-Server®

distributes in EventError events.• If you select true for the inbound-bsns-calls option, all established

inbound calls on a local agent are considered business calls.Monospace is also used for any text that users must manually enter during a configuration or installation procedure, or on a command line:

Example: • Enter exit on the command line.

User’s Guide 15

Preface Related Resources

Screen Captures Used in This Document

Screen captures from the product GUI (graphical user interface), as used in this document, may sometimes contain a minor spelling, capitalization, or grammatical error. The text accompanying and explaining the screen captures corrects such errors except when such a correction would prevent you from installing, configuring, or successfully using the product. For example, if the name of an option contains a usage error, the name would be presented exactly as it appears in the product GUI; the error would not be corrected in any accompanying text.

Square Brackets

Square brackets indicate that a particular parameter or value is optional within a logical argument, a command, or some programming syntax. That is, the parameter’s or value’s presence is not required to resolve the argument, command, or block of code. The user decides whether to include this optional information. Here is a sample:smcp_server -host [/flags]

Use of Angle Brackets

Angle brackets indicate a placeholder for a value that the user must specify. This might be a DN or port number specific to your enterprise. Here is a sample:smcp_server -host <confighost>

Note: In some cases (for example, VoiceXML and CCXML tags), angle brackets are required syntax elements and do not indicate placeholders. Notes in the text indicate where this is the case.

Related ResourcesConsult the following additional resources as necessary:• Genesys Voice Platform 8.0 Deployment Guide, which provides

information to install and configure GVP.• Genesys Voice Platform 8.0 VoiceXML 2.1 Help, which provides

information about developing VoiceXML applications. It presents VoiceXML concepts and provides examples that focus on the GVP implementation of VoiceXML.

• Genesys Voice Platform 8.0 CCXML Reference Manual, which provides information about developing CCXML applications for GVP.


Preface Related Resources

• Genesys Voice Platform 8.0 Troubleshooting Guide, which provides information about SNMP MIBs and traps for GVP, as well as troubleshooting methodology.

• Genesys Voice Platform 8.0 Configuration Options Reference, which replicates the metadata available in the Genesys provisioning GUI to provide information about all the GVP configuration options, including descriptions, syntax, valid values, and default values.

• Genesys Voice Platform 8.0 Metrics Reference, which provides information about all the GVP metrics (VoiceXML and CCXML application event logs), including descriptions, format, logging level, source component, and metric ID.

• Voice Platform Solution 8.0 Integration Guide, which provides information about integrating GVP 8.0 and SIP Server 7.6.

• Composer Voice 8.0 Deployment Guide, which provides installation and configuration instructions for Composer Voice.

• Composer Voice 8.0 Help, which provides online information about using Composer Voice, a GUI for the development of applications based on VoiceXML and CCXML.

• W3C Voice Extensible Markup Language (VoiceXML) 2.1, W3C Recommendation 19 June 2007, which is the W3C VoiceXML specification that GVP supports.

• W3C Voice Extensible Markup Language (VoiceXML) 2.0, W3C Recommendation 16 March 2004, which is the W3C VoiceXML specification that GVP supports.

• W3C Speech Synthesis Markup Language (SSML) Version 1.0, Recommendation 7 September 2004, which is the W3C SSML specification that GVP supports.

• W3C Voice Browser Call Control: CCXML Version 1.0, W3C Working Draft 29 June 2005, which is the W3C CCXML specification that GVP supports.

• Framework 8.0 Deployment Guide, which provides information to configure, install, start, and stop Framework components.

• Framework 8.0 Configuration Options Reference Manual, which provides descriptions of configuration options for Framework components.

• Framework 8.0 Genesys Administrator Help, which provides instructions for configuring and provisioning contact center objects using Genesys Administrator.

• Framework 7.6 SIP Server Deployment Guide, which provides information to configure and install SIP Server.

• Genesys Technical Publications Glossary, which ships on the Genesys Documentation Library DVD and which provides a comprehensive list of the Genesys and CTI terminology and acronyms used in this document.

User’s Guide 17

Preface Making Comments on This Document

• Release Notes and Product Advisories for this product, which are available on the Genesys Technical Support website at http://genesyslab.com/support.

Information about supported operating systems and third-party software is available on the Genesys Technical Support website in the following documents:• Genesys Supported Operating Systems and Databases• Genesys Supported Media InterfacesGenesys product documentation is available on the:• Genesys Technical Support website at http://genesyslab.com/support.• Genesys Documentation Library DVD, which you can order by e-mail

from Genesys Order Management at [email protected].

Making Comments on This DocumentIf you especially like or dislike anything about this document, please feel free to e-mail your comments to [email protected]. You can comment on what you regard as specific errors or omissions, and on the accuracy, organization, subject matter, or completeness of this document. Please limit your comments to the information in this document only and to the way in which the information is presented. Speak to Genesys Technical Support if you have suggestions about the product itself.When you send us comments, you grant Genesys a nonexclusive right to use or distribute your comments in any way it believes appropriate, without incurring any obligation to you.

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Preface Making Comments on This Document

User’s Guide 19

Part

1 OverviewThis part of the manual provides general information about Genesys Voice Platform (GVP) and the Genesys Administrator.This information appears in the following chapters:• Chapter 1, “Introduction,” on page 21• Chapter 2, “How GVP Works,” on page 25


User’s Guide 21

Chapter

1 IntroductionGenesys Voice Platform (GVP) is a software suite that integrates a combination of call processing, reporting, management, and application servers with Voice over IP (VoIP) networks, to deliver web-driven dialog and call control services to callers.This chapter introduces the GVP components and the Genesys Administrator, the GUI to configure and manage GVP.This chapter contains the following sections:

About GVP, page 21Genesys Administrator, page 23

About GVPThis section describes the GVP component applications and other objects in a GVP configuration:• GVP Components• IVR Profiles (see page 23)• GVP MIBs (see page 23)

GVP ComponentsGVP comprises the following components:• Resource Manager—Functions as a SIP Proxy that controls access and

routing to all resources in a GVP deployment. The Resource Manager is responsible for:

Allocating and monitoring resources.Managing sessions.Selecting services.Enforcing policies.


Chapter 1: Introduction About GVP

The Resource Manager also functions as a SIP Registrar, and monitors the health of GVP resources in the deployment.For information about how the Resource Manager performs its major functions, see “How the Resource Manager Works” on page 27.

• Media Control Platform—Provides media-centric services to other GVP components and to third-party gateways that use GVP services. The Media Control Platform is responsible for the execution of supported Voice Extensible Markup Language (VoiceXML) applications. Functions include:

Initiating, answering, transferring, and disconnecting calls.Playing audio and Text-to-Speech (TTS) prompts.Handling Automatic Speech Recognition (ASR) and DTMF inputs.Providing conference services.

For information about how the Media Control Platform performs its functions, see “How the Media Control Platform Works” on page 37.

• Call Control Platform—Provides call control capability in accordance with the supported W3C Call Control Extensible Markup Language (CCXML) standard. The Call Control Platform is optional in a GVP deployment. It operates as a SIP Back-to-Back User Agent (B2BUA) for requests to and from GVP components. Functions include:

Accepting, rejecting, and redirecting calls, including handling call setup information to enable intelligent routing and selective answering. Call-handling capabilities include supervised transfer, whispering, and call hold.Creating outbound calls through third-party gateways.Using Media Control Platform services to initiate VoiceXML dialogs, start conferences, and perform implicit transcoding.Providing multi-party conference support with moderator and floor control capabilities.Providing personal assistant services, such as dialing from a personal address book or corporate directory, managing personal appointments, and managing voicemail and e-mail.

For information about how the Call Control Platform performs its functions, see “How the Call Control Platform Works” on page 50.For information about creating CCXML applications to utilize Call Control Platform capabilities, see the Genesys Voice Platform 8.0 CCXML Reference Manual.

• Fetching Module, with Third Party Squid—Fetches content (such as audio files, VoiceXML pages, or CCXML pages) for the Media Control Platform and Call Control Platform sessions. Each Media Control Platform and Call Control Platform component in the deployment requires a Fetching Module. Third Party Squid is a caching proxy that the Fetching Module uses to improve response times by caching and reusing frequently requested resources.

User’s Guide 23

Chapter 1: Introduction Genesys Administrator

For information about how the Fetching Module and Squid perform their functions, see “How the Fetching Module Works” on page 55.

• Reporting Server—Stores and summarizes data and statistics submitted by Reporting Clients to provide near real-time reports by hour, day, week, and month. Reporting Clients on the Resource Manager, Media Control Platform, and Call Control Platform send call detail records (CDRs), Metrics, and Operational Reporting (OR) statistics to the Reporting Server. The Reporting Server provides an XML web services interface that is used by Genesys Administrator to obtain GVP reporting information. The XML web services interface is also accessible to any HTTP client, providing customers with access to GVP reporting outside of Genesys Administrator.For information about how the Reporting Server performs its functions, see “Logging and Reporting” on page 60.

For more general information about the GVP components, see the chapter about GVP architecture in the Genesys Voice Platform 8.0 Deployment Guide.

IVR ProfilesVoice Extensible Markup Language (VoiceXML) and Call Control Extensible Markup Language (CCXML) are the application-level languages that are used to construct voice and call control applications to control the interaction between the external user and the GVP software.Voice and call control applications are configured as IVR Profile objects in the Genesys Administrator UI. The IVR Profiles define how requests received by the VPS are translated into concrete service requests that can be executed by GVP components within the deployment.

GVP MIBsThe VP MIB Installation Package (IP) contains the Management Information Base (MIB) files that GVP uses to support Simple Network Management Protocol (SNMP).For general information about SNMP in a GVP deployment, see “SNMP Monitoring” on page 70. For detailed information about the MIBs, see the Genesys Voice Platform 8.0 Troubleshooting Guide.

Genesys AdministratorThe Genesys Administrator is a GUI that provides a web-based interface to the Genesys Configuration and Management Layers.Use the Genesys Administrator to deploy, configure, provision, and monitor GVP.Figure 1 shows a typical Genesys Administrator page.


Chapter 1: Introduction Genesys Administrator

Figure 1: Genesys Administrator

To access the Genesys Administrator for your Genesys deployment, go to the following URL:http://<Genesys Administrator host>/wcm

More Information

• For information about installing the Genesys Administrator, see the Framework 8.0 Deployment Guide.

• For general information about using the Genesys Administrator, see the online Framework 8.0 Genesys Administrator Help.

• For information about using the Genesys Administrator to configure and provision GVP Application objects and IVR Profiles, see Viewing or modifying GVP configuration parameters, page 78 and Configuring logical resource groups, page 127.

• For information about using the Genesys Administrator to monitor GVP and view reports, see Part 3 of this manual, starting on page 217.

TabsNavigation panel Details panel List selector(Options, Annex)View

Help

Tabs for selecting main functions—Monitoring, Provisioning, Deployment

User’s Guide 25

Chapter

2 How GVP WorksThis chapter describes how the Genesys Voice Platform (GVP) components operate in a GVP deployment. It also provides general information about the identifiers and Session Initiation Protocol (SIP) messages that are used in a GVP deployment.This chapter contains the following sections:

Sample Call Flow, page 25How the Resource Manager Works, page 27How the Media Control Platform Works, page 37How the Call Control Platform Works, page 50How the Fetching Module Works, page 55Logging and Reporting, page 60SNMP Monitoring, page 70Secure Communications, page 70GVP Identifiers and SIP Headers, page 71

Together with SIP Server 7.6 and Management Framework 8.0, GVP 8.0 constitutes the Voice Platform Solution (VPS) 8.0, which integrates voice self-service, agent-assisted service, and application management functions into a single, IP-based contact center solution.For more information about how GVP functions within a VPS deployment, see the Voice Platform Solution 8.0 Integration Guide.

Sample Call FlowGVP can be deployed and provisioned in a number of ways, to provide a range of services for inbound calls, outbound calls, transfers, and conferences.Figure 2 illustrates how GVP handles a typical inbound call.


Chapter 2: How GVP Works Sample Call Flow

Figure 2: Typical Inbound Call Flow

1. A call comes in to the SIP Server from an external source through a third-party media gateway.

2. The SIP Server passes the call to the GVP Resource Manager (SIP INVITE).3. The Resource Manager determines what to do with the call. If the

Resource Manager accepts the call, it matches the call to an IVR Profile, and selects a resource.For more information about how the Resource Manager selects services and resources and enforces policies, see “How the Resource Manager Works” on page 27.

4. The Resource Manager sends the call to a Media Control Platform or Call Control Platform resource (SIP INVITE). When it forwards requests to resources, the Resource Manager inserts additional SIP headers or parameters, as required by the service prerequisites, service parameters, and policies that have been configured for the IVR Profile. For more information, see “Service Request Modification” on page 32.

5. The Fetching Module for that Media Control Platform or Call Control Platform resource fetches the required VoiceXML or CCXML page from the application server (file, HTTP, or HTTPS request).

6. The Next Generation Interpreter (NGI) on the Media Control Platform or CCXML Interpreter (CCXMLI) on the Call Control Platform interprets the page and runs the application (VoiceXML or CCXML).

Genesys Voice Platform

SIPServer

Speech Server*

Application Server*

Web Services

User

GVP Reporting

Server

Composer Voice*GVP RS WS

Media Gateway

User Interface Layer

*Optional in VPS 8.0Note: Additional connections not shown here include :• TCP connections between GVP Reporting Server and RM , MCP, and CCP• Connections between Management Layer and all other Genesys components

MCP Fetching Module

and Squid

CCP Fetching Module

and Squid

1

8

Media Control Platform

6

7

5

5

7

Call Control

Platform*

6

9

9

2

Resource Manager

3 4

4

9

User’s Guide 27

Chapter 2: How GVP Works How the Resource Manager Works

7. Depending on the application, the Media Control Platform or Call Control Platform will request and use additional services:

For Automatic Speech Recognition (ASR) or Text-to-Speech (TTS) services, the Media Control Platform communicates with the third-party speech application server using Media Resource Control Protocol (MRCPv1 or MRCPv2).If the Call Control Platform requires conferencing or audio play/record services, it obtains them from a Media Control Platform resource.

The Media Control Platform or Call Control Platform send all requests for services from other GVP components through the Resource Manager (SIP or NETANN).

8. The Real-time Transport Protocol (RTP) media path is established between the Media Control Platform and the SIP end-user, in this example the originating caller through the media gateway.

9. The Resource Manager ends the call when one of the parties (the SIP end-user, the Media Control Platform, or the Call Control Platform) disconnects or when the call is transferred out of GVP (SIP BYE or REFER).

How the Resource Manager WorksThis section describes in more detail what the Resource Manager does with service requests, such as the call described in “Sample Call Flow” on page 25. The Resource Manager performs the following functions:• Session Management• Service Selection (see page 29)• Policy Enforcement (see page 32)• Service Request Modification (see page 32)• Resource Management (see page 32)• High Availability and Scalability (see page 37)For information about how the Resource Manager reports it activities to the Reporting Server, see “Logging and Reporting” on page 60.

Session ManagementA session is a set of related services that are used to deliver an end-user experience.There is a global logical session that encompasses the Resource Manager interactions with SIP Server. The global session is managed by SIP Server. Within the global session, the Resource Manager manages logical call sessions for specific GVP services, and individual call legs within a call session.



The Resource Manager manages GVP call sessions as follows:1. The Resource Manager creates a new call session when it receives a new

SIP INVITE request for a GVP service.2. The Resource Manager generates a GVP Session ID, and inserts this

information in the X-Genesys-GVP-Session-ID SIP extension header. For more information about session IDs, see “Session Identifiers” on page 72.

3. The Resource Manager maps the session to an IVR Profile (a voice or call control application) and identifies the type of service for each component session (call leg). For more information, see “Service Selection” on page 29.

4. The Resource Manager adds the following parameters to the X-Genesys-GVP-Session-ID header:

gvp.rm.datanode—To identify itself as the Resource Manager for the session. In GVP 8.0, the value of this parameter is not configurable.gvp.rm.tenant-id—To identify the voice or call control application under which the session executes. The value of this parameter is the name of the IVR Profile that was assigned when the profile was configured. For more information about IVR Profile IDs, see “Application Identifiers” on page 72.

5. The Resource Manager inserts the X-Genesys-GVP-Session-ID header when it forwards new SIP INVITE requests. The Resource Manager also adds the X-Genesys-GVP-Session-ID header when it forwards responses, if the header does not already exist in the response.

6. The Resource Manager inserts the X-Genesys-RM-Application-dbid header when it forwards the first SIP INVITE request to a GVP resource, to identify the IVR Profile under which the session executes. The value of this parameter is the Database Identifier assigned by Configuration Server (DBID) for the IVR Profile object. The GVP components need this information to log the IVR Profile DBID in the call detail records (CDRs) that they send to the Reporting Server.

7. When the Resource Manager receives a SIP INVITE request for a new call leg within an existing session (as identified by the X-Genesys-GVP-Session-ID header), the Resource Manager consults the policies of the IVR Profile and Tenant related to the existing session, to determine if the call leg can be created. For more information, see “Policy Enforcement” on page 32.If the Resource Manager cannot identify an existing session from the X-Genesys-GVP-Session-ID header (for example, because the session has timed out), the Resource Manager accepts and processes the incoming request without checking policies.

8. The Resource Manager maintains the session in accordance with configurable session inactivity and session expiry timers.The Resource Manager associates a session inactivity timer with each call leg, and monitors SIP traffic for the session to determine when a SIP

User’s Guide 29


session is stale. If the Resource Manager receives no SIP messages for the call leg within the inactivity interval, the Resource Manager internally cleans up the call leg data (application, tenant, and resource usage) as if a BYE was received.You can set session inactivity timers for each IVR Profile, for each tenant, for each resource, and for the Resource Manager. For more information about the session timer configuration parameters, see “Configuring Session Timers and Timeouts” on page 117.

Session-ExpiresHeader

The Resource Manager adds a Session-Expires header to initial INVITE requests if one is not present, and if the request does not contain the timer option in the Supported header. The value of the Session-Expires header is the configured value of the applicable session timer, except under the following conditions:

If the incoming request contains a Session-Expires header with a value greater than the configured value of the applicable session timer, and if the Min-SE header is also present, the Resource Manager reduces the value of the Session-Expires header to the greater of the Min-SE and configured session timer values.If the incoming request contains a Session-Expires header with a value greater than the configured value of the applicable session timer, but the Min-SE header is not present, the Resource Manager reduces the value of the Session-Expires header to the configured session timer value.If the incoming request contains a Session-Expires header with a value less than the minimum session expiry value configured for the Resource Manager (see proxy. sip.min_se on page 119), the Resource Manager rejects the request.If the incoming request contains a Session-Expires header with a valid value less than the configured value of the applicable session timer, the Resource Manager uses the value of the Session-Expires header.

The Resource Manager restarts the session inactivity timer each time it receives a SIP request or 200 OK response.

Service SelectionWhen the Resource Manager receives a request for a new SIP session, it maps the call to an IVR Profile, and then selects a service for the request. If the SIP request arrives in the context of an existing Resource Manager session for which a VoiceXML or CCXML application is already executing, the Resource Manager does not perform another mapping.



Mapping the Call to an IVR Profile

The Resource Manager maps the SIP request to an IVR Profile as follows:1. If the SIP Request-URI includes a gvp-tenant-id parameter, the Resource

Manager looks for an IVR Profile with a name that matches the value of the gvp-tenant-id parameter.

If the Resource Manager finds an IVR Profile that matches, it routes the SIP session to that application. When it does so, it removes the SIP Request-URI parameter from the outgoing request.If the Resource Manager does not find an IVR Profile that matches, it executes the default VoiceXML or CCXML application that has been configured for the Environment tenant (see default-application, in the gvp.general section, on page 154).

2. If the SIP Request-URI does not include a gvp-tenant-id parameter, but the VPS has been configured so that SIP Server provides DNIS information in the SIP header, the Resource Manager uses the DNIS that it extracts from the SIP message to map to an IVR Profile from a preconfigured DNIS resource list. When the Resource Manager routes the call to the application, it attaches the trunkport parameter to the SIP Request-URI, with the DNIS as the value.For information about configuring the mapping between DNIS ranges and IVR Profiles for the Environment tenant, see “Mapping IVR Profiles to Dialed Numbers” on page 151.A Resource Manager configuration option (sip-header-for-dnis, in the rm section) enables you to specify the header in which the Resource Manager will look for the DNIS. Ensure that the value you specify is consistent with the headers that you expect the Media Gateway to use. Valid values are:

The user part of the SIP Request-URI.The user part of the URI in the To header.The user part of the URI in the History-Info header, with index = 1.

For more information, see the description of the rm.sip-header-for-dnis configuration option on page 126.

Note: GVP 8.0 supports numeric DNIS only. All other characters (for example, *, #, a, b) are stripped from the incoming request.

3. If the Resource Manager cannot map the SIP request to an IVR Profile, the Resource Manager executes the new SIP session in the context of the default VoiceXML or CCXML application that has been configured for the Environment tenant (see default-application, in the gvp.general section, on page 154).

4. If the Resource Manager cannot map the SIP request to an IVR Profile and no default application has been configured for the Environment tenant, the incoming SIP request fails with a 404 Not Found response.

User’s Guide 31


Selecting the Service

After the IVR Profile for the Resource Manager session has been determined, the Resource Manager identifies the required service and the service prerequisites for each call leg.The Resource Manager performs service selection as follows:1. If the user part of the SIP Request-URI includes parameters that specify the

required service, the Resource Manager handles the SIP request as a request for the specified service. The Resource Manager appends service parameters to the Request-URI if they are not already included.Table 1 describes the parameters that the Resource Manager looks for in SIP messages, to specify the required service.

2. If the SIP Request-URI does not include parameters to identify the required service, and the request originates from a resource that supports voicexml or ccxml service, the Resource Manager handles the incoming SIP request as a request for external-sip service.

3. If the SIP Request-URI does not include parameters to identify the required service, and the request does not originate from a resource that supports voicexml or ccxml service, the Resource Manager uses the service type and

Table 1: Service Specified in SIP Request

SIP Request-URI Service Comment

User part is dialog, and contains a parameter with name voicexml

voicexml Service prerequisites included.

User part starts with dialog.vxml

voicexml Service prerequisites included.

User part is ccxml, and contains a parameter with name ccxml

ccxml Service prerequisites included.

User part starts with conf= conference Service prerequisites included. The remainder of the user part is the Conference ID for this request.

Contains a parameter user=phone

gateway Request originates from a resource that supports voicexml or ccxml service.

User part is composed of characters 0–9 and - (dash)

gateway Request originates from a resource that supports voicexml or ccxml service.



service prerequisites information that has been configured for the IVR Profile (see the configuration options described in “gvp.general Section” on page 141 and “gvp.service-prerequisite Section” on page 150).

4. If the Resource Manager cannot map the request to a service in accordance with the above rules, the Resource Manager rejects the request with a 404 Not Found SIP response.

Policy EnforcementThe Resource Manager tracks sessions and IVR Profile and service usage to enforce policies that are imposed on a per-application basis. The Resource Manager also consults dialing and service allowability rules to enforce policies that are imposed on a per-application and per-tenant basis.For the IVR Profile and Tenant configuration parameters that specify the policies you can configure for GVP, see “gvp.policy Section” on page 143 and “gvp.policy.dialing-rules” on page 148. Configuration options enable you to customize the SIP responses that are sent when the Resource Manager rejects a call because of policy criteria, as well as whether certain policy violations will trigger an alarm.

Service Request ModificationBefore the Resource Manager forwards the request to a resource that can handle the service, it adds, deletes, or otherwise modifies SIP parameters to capture user-defined data and to translate policy and other configuration information into SIP parameters, which can be extracted by the VoiceXML or CCXML applications. You can configure GVP so that different service parameters are used for each service of each application.For details about the service parameters and service prerequisites that you can configure for IVR Profiles and the Environment tenant, as well as the SIP parameters in which the Resource Manager captures this information, see “gvp.service-parameters Section” on page 149 and “gvp.service-prerequisite Section” on page 150.

External SIPService

External SIP service, which is one of the service types that GVP supports, enables the Resource Manager to route an outbound call through an external, RFC3261-compliant SIP proxy. When the Resource Manager forwards a request for external SIP service, it adds a Route header to the SIP message with a Route URI that includes the lr parameter.

Resource ManagementAll requests for GVP services go through the Resource Manager, which identifies a SIP resource capable of serving the request and forwards the request to it. The Resource Manager monitors GVP resources to maintain up-to-date status of the resources used in the GVP deployment. The Resource

User’s Guide 33


Manager manages GVP resources to provide load-balancing and, if applicable, High Availability for each resource type.The following subsections provide more details about how the Resource Manager performs its resource management functions.

Logical Resource Groups

The Resource Manager receives resource information from Genesys Management Framework. Logical resource objects in Management Framework represent groupings of resources that share common properties, such as service type (for example, voicexml), capabilities (for example, support for a specific VoiceXML grammar), and method of load-balancing (for example, round robin).Resources are grouped by the following service types:• voicexml

• ccxml

• conference

• gateway

• external

For detailed information about all the resource group properties, see Table 20 on page 131.Management Framework gives the Resource Manager a list of logical resource objects and a list of physical resources. Each physical resource belongs to a logical resource.

Monitoring Status

The Resource Manager acts as a SIP Registrar (RFC3261) for resources about which it receives information from Management Framework. The Resource Manager maintains information on the registration and usage status of each resource. If the logical resource group has been configured for monitoring (see monitor-method on page 132), the Resource Manager also monitors resource health. Health monitoring performed by the Resource Manager is separate from Simple Network Management Protocol (SNMP) management (see “SNMP Monitoring” on page 70).Configuration options enable you to control the monitoring behavior of the Resource Manager (see the registrar and monitor configuration sections for the Resource Manager Application object).

Selecting a Resource

ServiceCapabilities

After the Resource Manager has mapped a new SIP request to a service, it allocates the request to a logical resource group that can provide the service,



with the specific capabilities that are required by the VoiceXML or CCXML application.The Resource Manager uses two sources to identify what the capability requirements are:• The IVR Profile service policies (see “gvp.policy Section” on page 143).• Information parsed from SIP Request-URI parameters that have the prefix

gvp.rm.resource-req. The Resource Manager does not forward these Request-URI parameters.

Load-Balancing After the Resource Manager has selected a logical resource group for the service request, it allocates the request to a physical resource. Except for conference services (see “Selecting a Resource for Conference Services” on page 34), the Resource Manager selects the physical resource based on the load-balancing scheme for the group. Load-balancing options are:• Round robin—From a circular list, the Resource Manager chooses the next

resource whose usage has not exceeded configured limits.• Least used—The Resource Manager chooses the resource with the lowest

usage and whose usage has not exceeded configured limits. Usage is calculated as specified by the port-usage-type parameter (see port-usage-type on page 133).

Notes: The Resource Manager load-balances within a logical resource group. It does not load-balance between resource groups.The MRCP Client on the Media Control Platform, which provides Speech Resource Management (SRM), load-balances the selection of the third-party speech engines, on a round-robin basis.

Load-Balancingfor Gateway

Service

For gateway service, the Resource Manager selects a resource based on a configurable policy option that lets you specify whether the call must be routed to the gateway resource already associated with the session, or whether the usual load-balancing scheme will be used (see the IVR Profile gvp.policy.use-same-gateway configuration option, on page 148).

No ResourceSelected

If the Resource Manager is not able to select a resource to meet the request, the Resource Manager responds to the SIP request with a configurable error message (see “Customizing SIP Responses” on page 116 and Table 64 on page 279).

Selecting a Resource for Conference Services

Conference services have the special requirement that the Resource Manager must route requests for the same conference ID to the same conference resource, even if the requests come from different Resource Manager sessions.1. If the SIP Request-URI includes confmaxsize and confreserve parameters,

and the specified confmaxsize is less than the confreserve, the Resource Manager rejects the conference request.

User’s Guide 35


2. For the first request that the Resource Manager receives for a conference (in other words, the Resource Manager is not already handling requests with the requested conference ID), it identifies eligible conference resources by matching confmaxsize and confreserve requirements specified in SIP Request-URI parameters with conference maximums that have been configured for the IVR Profile and the resource group, in combination with its knowledge of the current status and usage of conference resources. For more information about the IVR Profile and resource group parameters that are considered, see “Enabling Conference Services” on page 102.

3. The Resource Manager selects a resource for the conference by load-balancing across the eligible resources, in accordance with the load-balancing scheme for the logical group (see “Load-Balancing” on page 34).The Resource Manager adds the confmaxsize and confreserve parameters to the outgoing Request-URI when it forwards the request.

4. When the conference session is successfully established, the Resource Manager increments the current usage of the resource by the expected size of the conference (as specified in the confreserve SIP Request-URI parameter—default value is 1 if the parameter was not defined).As new call legs join or leave the conference, the Resource Manager keeps track of the current conference size.

Note: Once the conference session is established, the maximum number of participants in the conference is the smallest of the conference size maximums (confmaxsize parameters) specified in the SIP request, the IVR Profile, and the resource group.

As a result, the Resource Manager may internally modify the confmaxsize parameter in the outgoing SIP Request-URI, and this might cause the Resource Manager to reject the conference request if the confmaxsize parameter in the outgoing SIP Request-URI becomes smaller than the confreserve parameter (see Step 1 on page 34)

5. When the Resource Manager receives subsequent requests with the same conference ID, the Resource Manager forwards the request to the same conference resource, provided the maximum conference size is not exceeded.If conference size maximums have not been defined in the SIP request, IVR Profile, or resource group, the Resource Manager forwards the request to the conference resource, and leaves it to the conference resource to reject the request if necessary.



If adding the new call leg means that the maximum size of the conference or the usage limit configured for the resource would be exceeded, the Resource Manager rejects the request.

6. If the Resource Manager is not able to select a resource to meet the request, the Resource Manager responds to the SIP request with a configurable error message (see “Customizing SIP Responses” on page 116 and Table 64 on page 279).

Failed Requests

The behavior of the Resource Manager in response to failed requests depends on the type of service and the failure response code received by the Resource Manager.• For voicexml, ccxml, and conference service requests for which it receives

a 4xx or 5xx response code, the Resource Manager tries to select another resource, in accordance with the load-balancing scheme for the group, until all resources in the group have been tried.

• For a gateway service request for which it receives a 4xx or 5xx response code or for which the request times out, the Resource Manager forwards the failure response to the User Agent Client (UAC).

• For any INVITE requests to create a new SIP dialog for which it receives a 6xx response, the Resource Manager immediately forwards the response to the UAC, without trying to select another resource.

• If the Resource Manager has received no successful 2xx responses and has received at least one final response from one of the resources, the Resource Manager chooses one of the received responses to forward to the UAC, in the following order of selection:

• If the Resource Manager has sent at least one request to a resource and has not received any final responses from any resource, the Resource Manager sends a 408 (Request Timeout) response to the UAC.

For more information about SIP response codes that are generated by GVP components, see “SIP Response Codes” on page 279.

Response Received Response Returned to the UAC6xx 6xx

401, 407, 415, 420, 484 401

Any other 4xx response The first 4xx response received

Any 5xx other than 503 The first 5xx response received

500 (Server Internal Error) 500 (Server Internal Error)

User’s Guide 37

Chapter 2: How GVP Works How the Media Control Platform Works

High Availability and ScalabilityGVP 8.0 supports GVP High Availability (HA) in two senses:• Resource components—Because resources are provisioned in logical

resource groups, the Resource Manager provides HA for GVP components in the normal way that it manages, monitors, and load-balances the resource groups. As long as more than one instance of a particular GVP Application has been provisioned in a resource group, the service that the GVP process provides will still be available to other VPS components if one of the provisioned instances is not available.

Resource ManagerHA

• Resource Manager—Windows Network Load Balancing (NLB) provides HA for the Resource Manager itself. You can configure two Resource Managers to run as a warm active-standby pair with a common virtual IP.Incoming IP traffic is load-balanced across NLB clusters. A Cluster Manager on each Resource Manager host monitors the Network Interface Cards (NICs) in the cluster, to determine when network errors occur. If any of the monitored NICs encounter an error, the Cluster Manager considers the network down. The load-balancing of incoming IP traffic is adjusted accordingly.For information about configuring the Resource Manager for HA, see “Enabling High Availability” on page 134.

Scalability NLB clustering also provides scalability, because adding Resource Manager hosts to a cluster increases the management capabilities and computing power of the Resource Manager function in the GVP deployment.

Note: GVP 8.0 does not support more than two Resource Managers in a cluster.

How the Media Control Platform WorksThis section provides information about the following topics, to explain how the Media Control Platform performs its role in a GVP deployment:• About the Media Control Platform (see page 38)• Operational Overview (see page 38)• Media Services (see page 42)• Speech Services (see page 44)• Transfers (see page 45)• Conferencing (see page 50)• Debugging VoiceXML Applications (see page 50)



About the Media Control PlatformThe Media Control Platform is composed of:• A core executable that consists of a number of application modules, such

as the Call Manager API (CMAPI) and the SIP Line Manager. (For a list of the Media Control Platform application modules, see “Media Control Platform” on page 255.)

• The Media Server, which is a group of libraries (libCMMPC.dll, libDSP*.dll, and third-party transcoder Dynamic Link Libraries [DLLs]) running in-process in the Media Control Platform, for media processing and Real Time Transport Protocol (RTP) streaming.

• The Next-Generation VoiceXML Interpreter (NGI), which is a DLL (libAPPVXML3.dll) running in-process in the Media Control Platform.

• The MRCP Client, which is a group of libraries (libmrcpclient.dll, libmrcpv1client.dll, libmrcpv2client.dll) running in-process in the Media Control Platform, to handle MRCPv1 or MRCPv2 communication with ASR and TTS speech engines.

The Fetching Module, which is a separate GVP component, co-resides on the Media Control Platform host. The NGI and the Fetching Module share memory.For more information about the Media Control Platform architecture and interfaces, see the chapter about GVP architecture in the Genesys Voice Platform 8.0 Deployment Guide.

Operational OverviewThe Media Control Platform receives requests for call and media services from the Resource Manager in the form of SIP INVITE messages. The platform can conference, transfer, or redirect calls using other kinds of SIP messages (see “Transfers” on page 45). The platform can also initiate outbound calls by sending SIP INVITE requests through the Resource Manager or directly to the destination.The platform provides media, conferencing, and other bridging services for both Media Control Platform and Call Control Platform calls. Media Control Platform services are defined by VoiceXML applications that are executed as part of establishing a SIP session between the platform and the service user. In addition, the platform supports NETANN conferencing services.

User’s Guide 39


Incoming Calls

The Media Control Platform handles incoming service requests for call or media services as follows:1. The Media Control Platform, acting as a SIP User Agent Server (UAS),

receives a SIP INVITE from the Resource Manager. Because the Resource Manager has modified the SIP request to insert service-prerequisites for the IVR Profile, the SIP Request-URI includes a voicexml parameter that specifies the URL of the initial page of the required VoiceXML application.Alternatively, the Media Control Platform may be configured to accept calls in which the originator specifies the initial VoiceXML URL in the Request-URI of the SIP INVITE (see sip.vxmlinvite on page 176). In these cases, provided the syntax and format of the Request-URI is correct, the normal Resource Manager method of mapping calls to IVR Profiles will be bypassed.

The Media Control Platform recognizes the following Request-URI formats:

sip:dialog.vxml.<URL>@host.com, where the URL portion must be properly encoded (draft-rosenberg-sip-vxml format)sip:<user>@host.com;voicexml=<URL> (NETANN dialog service format)sip:conf=<ID>@host.com (NETANN format, for calls to join a specified conference without going through a VoiceXML application)

The Media Control Platform supports the following service parameters in the Request-URI:

voicexml—Value must conform to the URI syntax defined in RFC 3986.maxage—Value must be all digits.maxstale—Value must be all digits.method—Value must be either get or post.postbody

timeout

gvp.alternatevoicexml—Specifies an alternative VoiceXML page if the NGI fails to fetch the primary page.gvp.config.<parameter name>—Sets the values of certain platform configuration options for the duration of the media session. This mechanism enables an IVR Profile to override certain Media Control Platform configuration parameters, for the session that is being executed in the context of that VoiceXML application. For the configuration parameters whose values can be set dynamically, see “Dynamic Media Control Platform Parameters” on page 294.



Special characters in the Request-URI parameters from the SIP interface must be URL-encoded. For example, characters that must be escaped include ? (%3F), = (%3D), and ; (%3B).The Resource Manager passes the value of the following IVR Profile gvp.policy parameters in the Request-URI, for handling by the Media Control Platform:

mcp-asr-usage-modemcp-max-log-levelmcp-sendrecv-enabled

For more information about these policy parameters, see Table 23 on page 143.If media services are required, the Resource Manager includes the SIP User Agent (UA) Session Description Protocol (SDP) offer in the SIP INVITE.

For more information about how the Media Control Platform negotiates media services, see Step 6 on page 41.

2. For valid INVITE requests, the platform immediately responds to the Resource Manager with a 100 TRYING message. In addition, configurable options enable you to specify whether the platform will also send intermediate provisional responses while the call is being set up.Provisional responses can include custom SIP headers, which must have the prefix, X-.

Note: The Media Control Platform does not support sending early media after a provisional response has been sent.

For responses that the Media Control Platform sends if an error occurs during call setup, see “SIP Response Codes” on page 279.

3. The Media Control Platform passes all the generic SIP Request-URI parameters to the NGI.

4. The NGI sends an HTTP/HTTPS or file retrieval request to the Fetching Module to fetch the initial page. The request includes the timeout, maxage, and maxstale values, if present, to determine if the fetch can be satisfied from the cache store.For more information about how caching is used to improve Media Control Platform performance, see “Caching” on page 56.

5. The NGI compiles and interprets the initial page, and all subsequent pages, for the Media Control Platform to run the application. The VoiceXML application is ready to proceed when the VoiceXML document is fetched, parsed, and compiled.The NGI supports the following encodings for VoiceXML pages and external ECMAScripts:

UTF-8

User’s Guide 41


UTF-16ISO-8859-xFar-East encoding for Japanese, Chinese and Korean

The NGI retrieves the encoding information for a document from the encoding attribute of the XML header or the charset attribute of the <script> tag.

6. At the same time that it passes SIP INVITE information to the NGI (Step 3), the Media Control Platform passes the SDP to the Media Server, to negotiate media capabilities.For information about the capability negotiation and the codecs that the Media Control Platform supports, see “Codec Negotiation” on page 43. For information about the file formats that are supported for audio and video play and record for various codecs, see “Audio and Video File Formats” on page 289.

7. The Media Control Platform sends a 200 OK response to the initial INVITE request when the VoiceXML application is ready to execute. The response includes the Media Server SDP answer, if applicable.If the initial INVITE and the ACK that is returned do not contain the required SDP information, a media-less dialog is established.In general, the VoiceXML application starts to execute when the 200 OK response is acknowledged (the platform receives an ACK). However, it is the VoiceXML application itself that decides whether or not it is ready to start execution. In particular, if a media-less dialog has been established, the VoiceXML application will not start executing until the platform receives a re-INVITE that includes the SDP information for the caller.

8. Once the VoiceXML application starts executing, it controls the session. The NGI is responsible for driving the Media Control Platform to appropriately execute the VoiceXML application. The NGI performs DTMF recognition, and issues commands to the platform to execute call and media operations.

The platform sends and receives SIP INFO messages for the following application events:

To accept DTMF digits—The SIP INFO content type is application/dtmf-relay, and content format is Signal = <digit>.

A configurable option (sip.sipinfodtmf) enables you to specify whether the application can also use SIP INFO messages to send DTMF.

The DTMF event is generated when the VoiceXML application tries to play an audio file that is named dtmf_<digit>.vox or dtmf_<digit>.wav.

To make a request—The application can specify the content type and content body of the SIP INFO message.



To send or receive data—The application can send data in custom SIP headers. The platform can send information that it receives in SIP INFO headers to the NGI, and this information is provided to the application in shadow variables.

The application uses dialogs to initiate transfers and conferences as required. For more information about how the Media Control Platform performs transfers, see “Transfers” on page 45.The NGI supports use of the userdata attribute on the <transfer> tag, to abstract computer-telephony integration (CTI) data. The NGI exposes CTI userdata to the application in a session variable, session.com.genesyslab.userdata.

The platform provides media services through the Media Server, for operations such as playing prompts and recording audio and video. For more information, see “Media Services”.For ASR or TTS, the Media Control Platform controls speech resources through the MRCP Client. For more information, see “Speech Services” on page 44.

9. The VoiceXML application can invoke other VoiceXML applications. The NGI is responsible for issuing commands to the Fetching Module to fetch VoiceXML pages and other applications.

10. When a caller disconnects (a BYE is received), the platform notifies the Voice XML application (through the connection.disconnect.hangup event). If the BYE includes a Reason header, the value of the Reason header is passed verbatim to the application.If the application disconnects, the platform generates a BYE request.

11. For each VoiceXML session, the Media Control Platform generates call detail records (CDRs), which it sends to the Reporting Server. For more information, see “CDR Reporting” on page 66.

12. For each VoiceXML session, the Media Control Platform sends logs and metrics (VoiceXML application event logs) to the log sinks, from where it sends them to the Reporting Server.For more information about metrics, see “Metrics” on page 64. For descriptions of the Media Call Control Platform metrics, see Genesys Voice Platform 8.0 Metrics Reference.

Media ServicesThe Media Server provides the following services:• Prompt playback• Utterance recording• DTMF collection• ASR streaming (streaming TTS audio to the SIP call, and streaming audio

data to an ASR server to perform speech recognition)

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• Audio and video encoding and transcodingThe media channel is established directly between the Media Server and the remote party (through a media gateway, if required), over RTP. The Media Server also supports Secure RTP (SRTP).

Selected Features

The Media Server provides advanced features for audio and video services. Features include:• Support for audio, video, and mixed audio-video for calls and conferences.• “VCR controls” that allow the caller to navigate within an audio or video

stream using DTMF keys (for example, pause, resume, and skip forward or backward).

• Full call recording for audio and video, including configurable support for recording DTMF input.

• Fine-grained control of conference input and output through configurable parameters for gain control, audio mixing, video switching, and so on.

• Mechanisms to guarantee the required level of real-time performance for time-critical functions (for example, generating output content in advance and buffering it).

• Per-prompt control of DTMF barge-in.• Support for Call Progress Analysis (CPA).

Codec Negotiation

The Media Control Platform supports the standard RFC 3264 offer/answer mechanism to negotiate capabilities for media services: The caller includes an SDP offer in the SIP INVITE, the receiving party answers with matched SDP capabilities in the 200 OK, and the originating caller acknowledges and confirms the negotiated SDP in an ACK message.The Media Control Platform also supports receiving SIP INVITE messages without SDP. In these cases, it generates an SDP offer in the 200 OK response. For outgoing calls, it also supports receiving SDP in the 183 Session Progress response.In addition, the platform supports in-call media information updates through a re-INVITE/200 OK/ACK sequence.



The Media Control Platform can support the following codecs:

A configurable parameter (mpc.codec) enables you to customize the list of codecs that are advertised in SDP offers, or that are used to match the remote party’s offer.If both the Media Control Platform and the remote endpoint are configured to negotiate multiple codecs for a call session, then multiple audio codecs may be used within a single SIP call.For information about the supported audio and video file formats, see “Audio and Video File Formats” on page 289.

Speech ServicesThe Media Control Platform manages MRCP Client sessions with third-party speech engines. The Media Control Platform provides speech recognition and speech synthesis commands to the MRCP Client, and the MRCP Client communicates these to the MRCP server(s) to carry out speech requests.• For MRCPv1, the MRCP Client uses RTSP to establish MRCPv1 control

sessions.• For MRCPv2, the MRCP Client uses SIP and SDP to create the

client/server dialog and set up the media channels to the server. It also uses SIP and SDP, over TCP or TLS, to establish MRCPv2 control sessions between the client and the server for each media processing resource required for that dialog.

The platform sends the RTP stream directly to the MRCP server for ASR, and receives the RTP stream directly from the MRCP server for TTS.

Grammars

The Media Control Platform generates the following grammars:• Hotkey grammars—Grammars that are used to match the UNIVERSALS

properties for the hotwords Help, Cancel, and Exit. There are separate grammar files for each supported speech engine. The hotkey grammars are installed in the <MCP Installation Path>\grammar\<engine name>\hotkey\ directory.

Note: The default hotkey grammars may not contain the correct strings for the hotwords in certain languages. Verify that the grammars are correct for the languages that are required in your deployment, and correct or add any strings that may be required.

• pcmu • g729 • h263 • telephone-event

• pcma • gsm • h263-1998

• g726 • amr • tfci

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• Builtin grammars—The set of builtin grammars provided in the VoiceXML specification. The Media Control Platform provides these because some engines do not support VoiceXML builtin grammars internally on the engine side. The builtin grammars are installed in the <MCP Installation Path>\grammar\<engine name> directory.

• Inline and Implied grammars—Menu and option grammars that are generated dynamically by the Media Control Platform. These grammars are temporarily stored in the <MCP Installation Path>\tmp directory.

In addition, the Media Control Platform supports native DTMF grammar handling with a builtin DTMF recognizer.For the default languages and builtin grammars that are supported when strict grammar mode is enabled, see the conformance.supported_* options in the vxmli configuration section.The Microsoft Internet Information Server (IIS) on the Media Control Platform host serves the grammars to the off-board ASR server. Ensure that you configure the IIS application server to serve the required grammars.

TransfersVoiceXML or CCXML applications use the <transfer> tag in VoiceXML dialogs to initiate transfers.

Transfer Types

From the point of view of the VoiceXML or CCXML application, there are three types of call transfers:• Blind—The application is detached from the incoming call (and the

outbound call if one is involved) as soon as the transfer is successfully initiated. This means that the application is unable to detect the result of the transfer request.

• Consultation (also referred to as supervised)—The application is detached from the incoming call when the transfer process finishes successfully. If the transfer process fails, the application retains a relationship with the call. This means that the application is able to report transfer failures.

• Bridge—The application does not detach from the incoming call, unless the incoming call disconnects. Control of the call always returns to the application when the transfer ends, regardless of the transfer result.

Whisper Transfer In addition, the whisper transfer feature enables the platform to delay connection of the caller and callee after the transfer operation has been performed.Whisper transfer enables the platform to continue performing media operations with the callee, and transfer the call out later. For the VoiceXML application developer, whisper transfer enables an application to be written to consult with the callee first, to determine whether the callee will accept the transferred call.



If the callee accepts the call, the transfer proceeds. If the callee rejects the call, the callee is disconnected, and the VoiceXML application can return control to the original caller.

Transfer Methods

To implement the requests for the different types of transfer at the telephony layer, the Media Control Platform can use the following SIP transfer methods:• HKF—Hookflash transfer, using DTMF digits (RFC 2833).

a. The Media Control Platform sends DTMF digits on the media channel. The platform leaves it to the media gateway or switch to perform the transfer on the network.

b. Configurable options enable you to specify whether the call will be disconnected by the platform or by the remote end. Otherwise, the call is disconnected after a configured timeout.

This is a one-leg transfer (in other words, it occupies only one channel on the platform).

• REFER—Transfer is based on a SIP REFER message (RFC 3515). a. The platform sends a REFER request to the caller, with the callee (as

specified in the VoiceXML application) in the Refer-To: header.b. The transfer fails if a non-2xx final response is received for the REFER.This is a one-leg transfer (in other words, it occupies only one channel on the platform).

• BRIDGE—The Media Control Platform bridges the media path.a. The platform sends an INVITE request to the callee, and a dialog is

established between the callee and the platform.b. The transfer fails if a non-2xx final response is received for the INVITE

request.This is a two-leg transfer (in other words, it occupies two channels on the platform). The platform stays in the signaling path and is responsible for bridging the two call legs.

• REFERJOIN—Consultative REFER transfer (RFC 3891). Also referred to as REFER with replaces transfer.a. The platform sends an INVITE request to the callee, and a dialog is

established between the callee and the platform.b. The platform also sends a REFER request to the caller, with the callee’s

information in the Replaces header.c. The platform considers the transfer to be successful if it receives a BYE

from the caller after a 2xx response for the REFER.d. The transfer fails if a non-2xx final response is received for the INVITE

request or for the REFER request.This is a two-leg, or join-style, transfer (in other words, it occupies two channels on the platform).

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• MEDIAREDIRECT—Media redirection transfer. The Media Control Platform uses SIP to handle call control between the caller and the callee, and the RTP media channel is connected directly between the caller and callee.a. The platform sends an INVITE request to the callee without SDP. b. If the transfer is proceeding, the callee responds with a 200 OK that

includes an SDP offer.c. The platform forwards the SDP offer in a re-INVITE request to the

caller.d. The caller responds with a 200 OK that includes the SDP answer.e. The platform forwards the SDP answer to the callee in an ACK response.f. The transfer fails if a non-2xx final response is received for the initial

INVITE request. This is a two-leg transfer (in other words, it occupies two channels on the platform).

The NGI controls the transfer method that is selected, based on the method attribute that is specified in the VoiceXML application. If the method is not specified, the default method for the applicable transfer type is used. The default methods are configurable (see sip.defaultblindxfer, sip.defaultconsultxfer, and sip.defaultbridgexfer, on page 167). In addition, configurable parameters enable you to specify whether the Media Control Platform will fall back to the BRIDGE or MEDIAREDIRECT method if one of the other methods fails.Because of the actual mechanisms involved, the SIP transfer methods do not all support all the transfer types. Table 2 summarizes SIP transfer method support for the different types of transfer.

Table 2: SIP Transfer Methods and Supported VoiceXML Transfer Types

SIP Transfer Method

Supported Transfer Type

Notes

HKF • Blind• ConsultationWhisper transfer supported

• The DTMF digits are flash or other configured digits, followed by a phone number.

• A different configured sequence of flash and digits can be dialed to abort the transfer.



REFER • Blind• Consultation

• The default platform method for type=blind.• Transfer connect timeout is not supported.• The platform can be configured to send an INVITE hold to the

caller.• For type=consultation, the platform also supports the NOTIFY method for notification of the transfer result. If the transfer fails, the platform puts the original caller off hold, and the VoiceXML application proceeds with the caller.

• The platform can be configured to send a BYE request to the caller, or to wait for a BYE from the caller.

• For transfer requests from the Call Control Platform, the Media Control Platform sends a REFER request to the Call Control Platform, which throws a dialog.transfer event to the CCXML application. The type attribute for the event is always set to blind, whether the request from the VoiceXML application was for blind transfer or consultation (supervised) transfer.

BRIDGE • Blind• Consultation• BridgeWhisper transfer supported

• The default platform method for type=bridge.• Non-whisper transfers support the connectwhen=immediate

attribute in the VoiceXML application. If this value is specified, a one-way media path from the callee to the caller is established before the call is connected.

• If specified in the VoiceXML application, the platform can continue to support media operations (such as handling DTMF grammars, ASR, transactional recording, and playing transfer audio) during a bridge transfer.

• For transfer requests involving the Call Control Platform, if the VoiceXML application uses a <send> tag to notify the Call Control Platform about a bridge transfer request, the Media Control Platform sends a SIP INFO message to the Call Control Platform.

Table 2: SIP Transfer Methods and Supported VoiceXML Transfer Types (Continued)

SIP Transfer Method


Notes

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Implications of Transfer Method–Transfer Type Combinations

In addition to varying levels of support for features such as whisper transfer and connection timeouts, the different combinations of SIP method and VoiceXML transfer type can result in scenarios that can have significant implications for metrics and general component activity logs and, therefore, for GVP reporting.

REFERJOIN • Blind• ConsultationWhisper transfer supported

• The default platform method for type=consultation.• The platform can be configured to send an INVITE hold to the

caller.• If the transfer fails, the platform puts the original caller off

hold, and the VoiceXML application proceeds with the caller.• The platform can be configured to send a BYE request to the

caller and then the callee, or to wait for a BYE from the caller.• For whisper transfer, if the callee rejects the transfer request,

the platform sends a BYE to the callee to disconnect the call.• Non-whisper transfers support the connectwhen=immediate

attribute in the VoiceXML application. If this value is specified, the media path is established between the caller and the callee as soon as the media session is ready.

MEDIAREDIRECT • Blind• Consultation• BridgeWhisper transfer supported

• For whisper transfer, a media channel is established between the callee and the Media Control Platform for the consultative part of the transfer, if necessary.

If the callee rejects the request, the platform sends a BYE request to the callee to disconnect the call. The media path and interaction between the platform and the caller then resumes.

• If the caller disconnects during the transfer (platform receives a BYE), the platform sends a BYE to the callee to disconnect the call.

• If the callee disconnects during the transfer, the platform updates the caller’s media path back to the platform, using a new re-INVITE if necessary.

Table 2: SIP Transfer Methods and Supported VoiceXML Transfer Types (Continued)

SIP Transfer Method


Notes


Chapter 2: How GVP Works How the Call Control Platform Works

ConferencingConferencing is, in essence, a special case of bridge-type transfer.Calls join a conference directly, by specifying the conference bridge identifier (conference ID) in the SIP Request-URI in NETANN format:sip.conf=<conf ID>@host.com

The Conference application module handles calls and manages call interactions with the conference bridge for NETANN conferencing. Platform-level configuration options (in the conference configuration section) support standard NETANN conference requirements, such as configurable participant roles (talk-only, listen-only, full duplex), audio gain parameters, and the video output algorithm (first, loudest, or no video).

Debugging VoiceXML ApplicationsThe NGI interfaces with the debug client GUI that is part of Genesys Composer Voice.If the real-time debugger is enabled (see vxmli.debug.enabled on page 178), information about calls is passed between the NGI and the debugger client in SIP INVITE and 18x messages.The debugger can skip or step through the NGI execution, execute JavaScript snippets, provide information about currently executing elements, and change some of the parameters for the elements being executed.The NGI can also save to the file system all the information related to the transactions of a call. This feature is helpful for debugging platform operations as well as VoiceXML applications.

How the Call Control Platform WorksThis section provides information about the following topics, to explain how the Call Control Platform performs its role in a GVP deployment:• About the Call Control Platform• Operational Overview (see page 51)

Incoming Connections (see page 51)Outgoing Connections (see page 53)

• Device Profiles (see page 53)

About the Call Control PlatformThe Call Control Platform is composed of:• A core executable (ccpccxml.exe), of which one of the main processes is

the HTTP event I/O processor (ioproc).

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• The CCXML Interpreter (CCXMLI), which is a static library (ccxmli.lib) running in-process in the Call Control Platform.

The Fetching Module, which is a separate GVP component, co-resides on the Call Control Platform host. The Call Control Platform and the Fetching Module share memory.

Operational OverviewThe Call Control Platform receives requests for call control or conference services for incoming connections from the Resource Manager in the form of SIP INVITE messages. The platform can conference, transfer, or redirect calls using other kinds of SIP messages (see “Transfers” on page 45). The platform can also initiate outbound calls by sending SIP INVITE requests through the Resource Manager or directly to the destination.For more detailed information, see the Genesys Voice Platform 8.0 CCXML Reference Manual.

Incoming Connections

The Call Control Platform handles service requests for incoming connections as follows:1. The Call Control Platform receives a SIP INVITE from the Resource

Manager.2. The Call Control Platform assigns a device profile to the connection. For

more information about device profiles, see “Device Profiles” on page 53.3. The Call Control Platform can accept, reject, or redirect the connection.

Configuration options allow you to customize some of the SIP responses that the Call Control Platform sends to the Resource Manager for the respective events (see “Customizing SIP Responses” on page 116 and Table 64 on page 279).

4. For connections that it accepts, the Call Control Platform sends an HTTP/HTTPS or file retrieval request to the Fetching Module to fetch the initial page.

Because the Resource Manager has modified the SIP request to insert service-prerequisites for the IVR Profile, the SIP Request-URI includes a ccxml parameter that specifies the URL of the initial page of the required CCXML application.If the Request-URI from the Resource Manager does not include an initial page URI, or if the Call Control Platform is being used in a deployment without the Resource Manager, the Call Control Platform uses the default that has been configured for the platform (ccpccxml.default_uri).



Call Control Platform HTTP requests comply with Hypertext Transfer Protocol (HTTP) 1.1. For secure HTTP (HTTPS), the Call Control Platform supports HTTP over Secure Socket Layer (SSL) 3.0 and HTTP over Transport Layer Security (TLS).The HTTP/HTTPS fetch method (get or post) depends on whether the method parameter was specified in the SIP Request-URI. The default is get.

Other parameters that are supported in the HTTP/HTTPS fetch are:namelist—A list of ECMAScript variables whose values are submitted as part of the request.enctype—The encoding type to be used for namelist data, if the method is post. The only supported value is application/x-www-form-urlencoded.

For both get and post methods, namelist variables must be encoded in the URI query string in the form url-encoded format (as described in the HTML 4.01 specification). If the get method is used, the namelist variables are appended after a ? (question mark) character.

5. The CCXMLI compiles and interprets the initial page, and all subsequent pages, for the Call Control Platform to run the application.

A CCXML page may transition to another page as it executes, but only one page is executed at a time.The CCXML session may create and interact with other entities:

Connections (other SIP sessions)Dialogs (VoiceXML sessions)ConferencesOther CCXML sessions

To improve performance, the Call Control Platform enables the root page of the initial page to be cached. Caching is not relevant for the initial page itself, or for other pages, because CCXML application pages are session-specific. For more information about how the Fetching Module caches pages, see “Caching” on page 56.The Call Control Platform supports receiving DTMF events in SIP INFO messages, and propagates the events and data to the CCXML application and other connections.

6. The Call Control Platform uses the Media Control Platform to provide bridging, conference, and transcoding services. It may also perform implicit conferencing and transcoding if the endpoints of a connection do not have the required bridging capabilities or support required codecs. The Call Control Platform obtains these services by sending SIP requests through the Resource Manager.Dialog-initiated transfers between CCXML sessions, or between CCXML and VoiceXML sessions, are application driven, with the SIP messaging going through the Resource Manager in SIP INFO messages. For more information about transfers, see “Transfers” on page 45.

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7. For each CCXML session, the Call Control Platform generates call detail records (CDRs), which it sends to the Reporting Server. For information about the CDR attributes, see “CDR Reporting” on page 66.

8. For each CCXML session, the Call Control Platform sends logs and metrics (CCXML application event logs) to the log sinks, from where it sends them to the Reporting Server.For more information about metrics, see “Metrics” on page 64. For descriptions of the Call Control Platform metrics, see Genesys Voice Platform 8.0 Metrics Reference.

Note: The Call Control Platform does not support operational reporting (OR).

9. If configured to do so (see ccxmli.debug.data.* and ccxmli.platform.save.* parameters), the Call Control Platform captures fetch data for CCXML and ECMAScript files, to aid in debugging CCXML applications.

Outgoing Connections

The Call Control Platform can start a new CCXML session if it receives a session creation request directly from a web server, to place an outbound call. Alternatively, the Call Control Platform may place an outbound call within the context of an existing session.The CCXML application uses the <createcall> tag to create the connection, and specifies the destination of the call (in the dest attribute) as a SIP URI. The value of the dest attribute is used in the To: header of the SIP INVITE that the Call Control Platform sends to the Resource Manager to place the call. The Resource Manager, in turn, forwards the request to another SIP Proxy that may have been configured in a routeset for the Call Control Platform, or to the SIP Server.

Note: You can override the default outbound proxy configured in the route set by specifying an outboundproxy hint in the CCXML <createcall> tag.

Device ProfilesThe Call Control Platform interacts with a variety of SIP devices, all of which have different characteristics and features. The concept of a device profile enables the Call Control Platform to interact with a wide range of devices, even though they might differ in the way that they support SIP.The device profile defines a number of properties that describe the SIP and SDP capabilities of a class of devices. The Call Control Platform uses a device profile when it performs call control operations (for example, <join> and



<accept>). The Call Control Platform assigns a device profile to any SIP device with which it interacts. The properties of the device profile then govern how the Call Control Platform interacts with the SIP device.

Device ProfileConfiguration File

The properties of the various device profiles are defined in a text file in the Call Control Platform config directory (<Call Control Platform Installation Directory>\config\ccpccxml_provision.dat). For more information about the properties that are defined for CCXML device profiles, see Table 33 on page 189.

Assigning DeviceProfiles

The Call Control Platform assigns device profiles as follows:• Incoming connections—The Call Control Platform tries to match the SIP

header from the incoming SIP INVITE with the value of the SIP Header Name property that is defined in the device profile configuration file, by order of precedence that is also specified in the configuration file. (By default, the SIP header that the platform looks for is User-Agent.) If it cannot match the SIP header, the Call Control Platform uses the Default Inbound profile that has been provisioned (see “Default Device Profiles”).

• Outbound connections, dialogs, and conferences—The Call Control Platform matches CCXML hints with the value of the Device Profile Name property that is defined in the device profile configuration file. If it cannot match the hint, the Call Control Platform uses the Default Outbound, Default Dialog, or Default Conference profile that has been provisioned (see “Default Device Profiles”).

Default DeviceProfiles

By default, VP Call Control Platform 8.0 is provisioned with the following device profiles for SIP devices, by order of precedence:• Cisco Gateway• Audiocodes Gateway• Convedia Media Server• X-Lite• Brooktrout Snowshore• GVP MCP• Audiocodes MP 104• Default Inbound• Default Outbound• Default Conference• Default DialogFor the property values that have been defined for the preprovisioned device profiles, see Appendix D on page 304.If your deployment uses SIP devices that are not adequately represented by the default device profiles, you must provision additional device profiles or else modify an existing device profile. For more information, see “Configuring Device Profiles” on page 188.

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Chapter 2: How GVP Works How the Fetching Module Works

How the Fetching Module WorksThis section provides information about the following topics, to explain how the Fetching Module and the Squid Caching Proxy perform their role in a GVP deployment:• About the Fetching Module• Caching (see page 56)

Non-HTTP/1.1-Compliant Caching (see page 56)HTTP/1.1-Compliant Caching (see page 56)Squid Configuration File (see page 59)Squid Log Files (see page 60)Managing the Cache Manually (see page 60)

About the Fetching ModuleThe Fetching Module is an executable (pwproxy.exe) that is responsible for fetching VoiceXML and CCXML file and HTTP/HTTPS resources.The Fetching Module uses shared memory to communicate with the NGI (on the Media Control Platform host) or CCXMLI (on the Call Control Platform host). Therefore, even though they are separate processes, the Fetching Module can pass fetch results and other information back to the interpreters very efficiently.

Note: The Fetching Module must be started before the Media Control Platform and the Call Control Platform are started. If the Fetching Module stops for any reason, the Media Control Platform and the Call Control Platform must also be stopped, then restarted after the Fetching Module has restarted.

Fetching ModuleCaching

To improve performance, the Fetching Module performs caching as follows:• The Fetching Module itself performs some limited in-memory caching,

which is not HTTP/1.1 compliant.• If the Fetching Module determines that it cannot serve the request from its

in-memory cache, it goes to the Squid Caching Proxy to try to fetch the content. The Squid Caching Proxy performs HTTP/1.1-compliant caching.

• If Squid determines that it cannot serve the content from its cache, it goes to the Web Server to try to fetch the content.

For more information, see “Caching”.



CachingAudio and video recordings are common in VoiceXML documents, and they can be very large. Their content is also mostly static. Therefore, using cached content significantly improves performance.Unlike visual browsers, there are no end-user controls in the VoiceXML interpreter context to enable stale content to be updated or refreshed. Rather, the VoiceXML document itself enforces cache refresh, through appropriate use of the maxage and maxstale attributes. However, these attributes interact with other proxy settings and HTTP cache-control mechanisms at various levels, as described in the following subsections.

Non-HTTP/1.1-Compliant Caching

The Fetching Module caches documents in-memory, in accordance with configurable maximum age and URL substring parameters (see the iproxy.* parameters described in “Important Fetching Module Configuration Options” on page 198). For the algorithm that the Fetching Module uses to determine whether it will use a cached version in its own memory, see “Fetching Module Caching Algorithm” on page 315.Because this level of caching is non-compliant with HTTP/1.1, use it carefully. If you require strict HTTP/1.1 compliance in your deployment, set the iproxy.* parameters so that this in-memory caching is turned off.

HTTP/1.1-Compliant Caching

The Fetching Module uses a caching proxy (Third-Party Squid) for HTTP/1.1-compliant caching. The caching proxy generates HTTP/1.0 requests, but supports HTTP/1.1 caching functionality.The caching policies of the VoiceXML interpreter context adhere to the cache correctness rules of HTTP/1.1. In particular, the Expires and Cache-Control headers are honored.

Caching Policies

• The application server maintainer/content provider can provide guidelines for content expiry using the Cache-Control and Expires HTTP response headers.

• If these headers are not present, Squid will use heuristics to generate expiry times.

• The application developer can deterministically control the caching behavior of application resources, by using the maxage and maxstale attributes for each URI-related VoiceXML tag. This behavior includes forcing a validation of the current cache contents (using maxage), and accepting expired cache contents (using maxstale).

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• The platform maintainer can control cache resource usage by the way in which Squid is configured.

Caching Behavior

The primary impact of the caching policies is that the client has control over what it will accept from the cache, even if the server has specified an Expires header or maxage/maxstale attributes, or if the caching proxy has generated an expiry time itself.• Documents from the web server will be delivered with zero, one, or both of

the response headers.• If an Expires header is present, it is used to set the expiry time of the object

in the cache. • If the Expires header is not present, Squid applies a heuristic to set an

expiry time. • If a Cache-Control header is in the response, it will be used to control

expiry times, and will override an Expiry time if also provided.For the algorithm that Squid uses to determine whether it will fetch a fresh version, see “Squid Caching Algorithm” on page 316.

Note: It is an optimization to perform a get if modified (the request includes an If-Modified-Since [IMS] header) on a document still present in the cache when the policy requires a fetch from the server. Squid does perform this optimization.

Maxage and maxstale

VoiceXML enables the application developer to control caching policy for each use of each resource.The application developer can specify maxage and maxstale attributes for each resource-related element. These attributes provide fine-grained control over when documents are returned from the cache, and when they are fetched from the origin server. For example:• Setting maxage to a non-zero value means that Squid might be forced to get

a fresh copy of a resource that may not yet have expired in the cache. Setting maxage to zero means that Squid will unconditionally be forced to get a fresh copy.

• Using maxstale enables the application developer to state that an expired copy of a resource that is not too stale (according to the rules of HTTP/1.1) may be used. This can improve performance by eliminating a fetch that would otherwise be required to get a fresh copy. This is especially useful for application developers who might not have direct server-side control of the expiration dates of large static files.



Notes: Like other caching proxies that support maxage and maxstale, Squid does not delete items from the cache after their expiry time, unless other cache requirements (such as memory or disk usage limits) dictate such action. The reason for this is that the client may specify that an expired resource is acceptable.Some resources may be addressed by URIs that name protocols other than HTTP and that do not support the maxage and maxstale attributes. If the protocol does not support the notion of resource age, the interpreter context computes the age of a resource from the time it was received. If the protocol does not support the notion of resource staleness, the interpreter context considers the resource to have expired immediately upon receipt.

The maxage and maxstale attributes interact with server-provided expiry times to produce a variety of caching behaviors. Table 3 describes some sample behaviors.

Maxage, Maxstale, and the Initial Page

For the initial page request, the GVP Session ID is submitted as part of the URL. Because this ID is unique, the requested URL appears unique, so the maxage and maxstale parameters have no meaning for that page. However, they do have meaning for the initial root page.Configuration parameters in the Media Control Platform set the values of the maxage and maxstale parameters for the initial root page (see vxmli.initial_request_maxage and vxmli.initial_request_maxstale). For both parameters, the default value is -1 (undefined).

Table 3: Using maxage and maxstale Attributes

Desired Behavior maxage maxstale Notes

Client control over expiry

<desired_expiry> 0 • Caching based on Expires header.• Refetch based on maxage and maxstale.

• Uses IMS.

Expired document acceptable

<large_value> <desired_maxstale> • Caching based on Expires header.• Refetch after Expiry time plus maxstale.

• Uses IMS.

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Determining Expiry Time

Web servers may or may not return an Expires response header to the client.• If the Web Server does return an Expires response header, this expiry time

is used in the cache refresh algorithm.• If, instead, the Web Server provides expiry information as part of a Cache-

Control header (using maxage/maxstale), this information will be used to control cache expiry.

Expiration Model Squid uses a Refresh-Rate model, rather than a time-based expiration model. Objects are not purged from the cache when they expire. Instead of assigning a “time-to-live” when the object enters the cache, Squid checks freshness requirements when objects are requested.• If an object is “fresh”, Squid gives it directly to the client.• If an object is “stale”, Squid makes an If-Modified-Since request for it to

the Web Server.For the algorithm that Squid uses to determine freshness or staleness, see “Squid Expiry Time Algorithm” on page 316.

Squid Configuration File

The Squid configuration file (C:\squid\etc\squid.conf) controls configuration of the caching proxy. The configuration file is a text file that contains pairs of keywords and values (with no equals [=] sign). For example:http_port 3128

defines port 3128 as the TCP port that the caching proxy will use for receiving requests.In general, the default Squid configuration file should be suitable for most installations. However, you might need to modify the Squid configuration file for the following reasons:• You need to configure for a second-level proxy.• You cannot configure your Web Server to deliver Expires headers, and you

wish to change the Squid defaults for the expressions Squid tries to match in SIP request-URI headers to control refresh behavior.

• You need to configure non-standard “safe” ports or SSL ports for HTTP and SSL.

For more information about modifying the Squid configuration file, see “Configuring the Squid Caching Proxy” on page 200.For details about all Squid configuration items, see the Squid Configuration Guide (http://squid.visolve.com/squid24s1/contents.htm).Changes to the Squid configuration file are not reflected in the running configuration immediately.


Chapter 2: How GVP Works Logging and Reporting

Squid Log Files

The caching proxy logs can provide useful information to assist you in identifying performance issues or resolving VoiceXML or CCXML application problems.

Access Logs

The Squid access.log file is in the following location: C:\squid\var\logs\

The access log contains one entry for each HTTP (client) request and each Inter-Cache Protocol (ICP) Query. HTTP requests are logged when the client socket is closed. The native access.log has ten fields. A single dash (-) indicates unavailable data.For details about the fields in the Squid access.log, see “Squid Access Logs” on page 317.

Managing the Cache Manually

Table 4 summarizes the commands to force refreshes of cached objects, purge an object from the cache, or clear the entire cache. Issue these commands in the cmd console window on the Media Control Platform or Call Control Platform host whose cache you want to manage.

Logging and ReportingEMS Reporting refers to the GVP logging and reporting feature, which provides the following services:• Accumulates key measurements and data describing the calls being

processed by the deployment.

Table 4: Manual Cache Management Commands

Objective Command

Refresh an object in the cache. C:\squid\bin\squidclient -r <uri>

where <uri> is the full URI of the object you want to refresh.

Purge an object in the cache. C:\squid\bin\squidclient -m PURGE <uri>

where <uri> is the full URI of the object you want to purge.

Clear the entire cache. C:\squid\bin\sbin\squid -k shutdown -n SquidNTecho '' > C:\squid\var\cache\swap.statenet start SquidNT

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• Delivers infrastructure for reliably delivering data to a relational backend.• Provides services for near real-time reporting on operational aspects of the

deployment.• Provides historical reporting on VoiceXML and CCXML application

usage.EMS Reporting leverages the Management Framework role-based security system to offer consistent, permissions-based access to data.This section provides information about the following EMS Reporting topics:• EMS Reporting Architecture• EMS Logging (see page 63)

Logs (see page 63)Metrics (see page 64)Log Sinks (see page 65)

• CDR Reporting (see page 66)• OR Service (see page 68)• Reporting Client (see page 68)• Reporting Server (see page 68)• Reporting Web Services (see page 69)



EMS Reporting ArchitectureEMS Reporting uses a client/server architecture. Figure 3 illustrates the EMS Reporting architecture.

Figure 3: EMS Reporting Architecture

EMS Logging • An EMS Logging interface on each component, or GVP Application, enables the component to log events relating to component activity. For more information, see “EMS Logging” on page 63.

• Additional interfaces on each call-processing component provide the following services to accumulate data:

CDR Reporting Call Detail Record (CDR) Service, which enables the component to submit and update CDRs to the Reporting Server. For more information about CDRs, see “CDR Reporting” on page 66.

OR Service Operational Reporting (OR) Service, which accumulates call arrival, call length, and call peak statistics. (Applicable for the Resource Manager and Media Control Platform only.) For more information about OR reporting, see “OR Service” on page 68.

The call-processing components are the Resource Manager, Media Control Platform, and Call Control Platform Applications.

Reporting Server

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MS SQL 2005 /Oracle 10g

Call Processing ServerComponent

EMS Logging

Reporting Client

Log Files

Op. Reporting

CDR Service

Management Framework

Config Server / Message Server

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DB MaintenanceProcess

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VAR • A <log> tag interface supports the Voice Application Reporter (VAR) reporting product, which is delivered by the Reporting Server. The <log> tag interface enables users to demarcate their VoiceXML applications into logical transactions, and assign success or failure to individual transactions or to a call as a whole. The <log> interface also provides a means of attaching application-specific data (such as call notes and custom name-value pairs) to calls.The Reporting Server accumulates summary statistics based on the processing of appropriately formatted VAR <log> tags. The summary statistics that it derives are accessible through web services and the Genesys Administrator.For more information, see “VAR Metrics” on page 65 and Table 73 on page 301.

Reporting Client • A Reporting Client on each call-processing component is responsible for reliably delivering the accumulated data to the Reporting Server. In GVP 8.0, the data is transported over TCP. For more information, see “Reporting Client” on page 68.

Reporting Server • The Reporting Server receives and persists data submitted from the Reporting Clients on the call-processing servers. There can be only one Reporting Server in a GVP deployment. For more information about the Reporting Server, see “Reporting Server” on page 68.

EMS LoggingThe EMS Logging API enables GVP components to raise logging events at two levels:• At the level of the component, or GVP Application—these are referred to

as logs.• At the level of the VoiceXML or CCXML application—these are referred

to as metrics.

Logs

Logs include three important elements by which they can be filtered: Severity, Module ID, and Specifier.

Severity GVP components, like most other Genesys components, can raise events at the following levels of severity (in descending order of severity):0 = Critical1 = Error2 = Warning3 = Note



4 = Info5 = Debug

Module IDs Each GVP component is composed of one or more Application Modules, each of which is assigned a Module ID. The component logically organizes the logs it emits by Module ID.

Specifiers A specifier is a number that uniquely identifies a given event that is logged by a given module.For a list of Module IDs and specifiers that are used in GVP 8.0, see Appendix A, “Module and Specifier IDs,” on page 255.

Additional LogData

EMS Logging associates a UTC timestamp, to millisecond precision, with each logging event when it is raised.Logs for call-processing component events that are associated with a call session include the GVP Component ID and the GVP Session ID, as well as the UTC timestamp. Metrics can therefore be mapped to CDRs, which provide further information, such as call start time, call end time, the Application ID (DBID of the IVR Profile), and local and remote SIP URIs.For more information about GVP IDs, see “GVP Identifiers and SIP Headers” on page 71.

Log Delivery Log events are delivered to one or more log sinks for the component (see “Log Sinks”), and then sent on to Management Framework or the Reporting Server.By default, log files are located in the following folders:C:\Program Files\GCTI\gvp\<IP name>\<Your component application

name>\logs\

Metrics

Metrics describe application-level events, and have no severity.Each metric has a unique type identifier (for example, start_session) and is associated with a specific VoiceXML or CCXML session ID. The body of the metric is defined by the component. For example, the body of a metric can be a text string that consists of a number of pipe-delimited parameters (such as ANI|DNIS|SIP Request-URI), encoded in UTF-8.

Metrics Examples The following are examples of the kinds of metrics that are logged. For full details about the metrics that are available in GVP 8.0, see the Genesys Voice Platform 8.0 Metrics Reference.• The Media Control Platform logs an INCALL_BEGIN metric when an inbound

call is accepted.• The NGI logs a PROMPT metric when it starts to play back a prompt queue.• The time to fetch a VoiceXML page is measured and logged.

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VAR Metrics VAR metrics are events that are generated by the Media Control Platform when it encounters VAR-specific <log> tags in the VoiceXML applications. The VAR-specific <log> tags have the prefix com.genesyslab.var.For the metrics that the Media Control Platform generates when the NGI executes a VAR-specific <log> tag, see Table 73 on page 301.For more information about using <log> elements in VoiceXML applications, see the Genesys Voice Platform 8.0 VoiceXML 2.1 Help.

Metrics Delivery Metrics are delivered to the log sinks for the component (see “Log Sinks”). Upstream metrics, also referred to as call events, are metrics that are configured to be sent to the Data Collection Sink (DATAC) and then sent on to the Reporting Server for storage and reporting purposes. The Media Control Platform and the Call Control Platform are the only sources of upstream metrics.

Log Sinks

Every component that uses logging has configurable access to one or more log sinks, which receive a real-time stream of logs or metrics, as defined by configurable filters (see ems.logconfig.<Sink Name> on page 106 and ems.metricsconfig.<Sink Name> on page 107).The log sinks are DLLs that are loaded dynamically at runtime. For the default log sinks that are attached to each component, see ems.log_sinks on page 105.The log sinks enable EMS Reporting to implement upstream reporting, integrate with Management Framework, and accumulate summary statistics that are used by the Reporting Server. Upstream reporting refers to the ability of components to send a configured subset of metrics to the Reporting Service for storage and reporting purposes.The following log sinks are available in GVP:• MFSINK—The Management Framework Adaptation Sink. MFSINK connects

the GVP and Management Framework logging systems, through CCILib, for file- and network-based logging. Configurable parameters in the log configuration section for each GVP component determine what Management Framework does with the logs—for example, writing them to file or delivering them to Message Server.

• DATAC—The Data Collection Sink. DATAC derives resource-specific summary statistics, and delivers summary statistics and metrics to the Reporting Server, where they can be queried through the Call Events reporting service. (Not applicable for the Resource Manager or Fetching Module.)A configurable option on the Media Control Platform and Call Control Platform (ems.dc.default.metricsfilter) enables you to specify which of the metrics delivered to DATAC will be forwarded to the Reporting Server.



• TRAPSINK—The SNMP integration sink. For GVP components that have been configured to raise traps, TRAPSINK forwards log messages to the Management Data Agent library, which the GVP process uses to implement the applicable MIBs.

Depending on the configured filters, a particular log or metric may be directed to more than one log sink for the component, or to none. If a given log event does not match the configured event types for that component’s log sinks, the log event is silently discarded.The log sinks themselves may generate log events, and can therefore have one or more log sinks attached to them. For example, in GVP 8.0, DATAC has an MFSINK, which enables DATAC logs to be delivered to Message Server or to file.

Calculated Statistics

The data in DATAC includes statistics that are calculated from metrics data.

CDR ReportingCDRs are records that describe key attributes of a call session that is being, or has been, processed by the deployment.The CDR Service on the Resource Manager, Media Control Platform, and Call Control Platform enables the component to submit and update CDRs to the Reporting Server in near real-time.The Reporting Server correlates the CDRs based on the GVP Session-ID (see “Session Identifiers” on page 72).The intervals at which the Reporting Client submits CDRs to the Reporting Server depends on configuration (see ems.rc.cdr.batch_size on page 108).

CDR Attributes

The CDRs share a common set of attributes that, at a minimum, the component must include. In addition, the Media Control Platform and Call Control Platform include certain attributes that are specific to the component type.

Common CDR Attributes

All components include the following attributes in the CDRs that they submit:• Session start and end time.• IDs for the VoiceXML or CCXML application; Media Control Platform or

Call Control Platform session; Resource Manager session; and overall Genesys session (UUID).

• Call type—Available types are:Inbound (1)—for Resource Manager and Media Control PlatformOutbound (2)—for Resource Manager and Media Control Platform

User’s Guide 67


Bridged (3)—for Media Control PlatformUnknown (4)—for Resource ManagerNew Call (5)—for Call Control PlatformCreate-CCXML (6)—for Call Control PlatformExternal (7)—for Call Control Platform

• Local-URI—The URI that identifies the local service that was delivered.• Remote-URI—The URI of the party with whom the dialog was conducted.

The platform obtains this information from the From header on an inbound call or the Request-URI on an outbound call.

Media Control Platform–Specific Attributes

The Media Control Platform includes the following additional attribute in CDRs:• For bridged calls, the parent Component ID (in other words, the Media

Control Platform ID of the call session that originated the bridged session).

Call Control Platform–Specific CDR Attributes

The Call Control Platform includes the following additional attributes in CDRs:• How the session was started:

EXTERNAL—The session was created through the HTTP session creation I/O processor.CREATECCXML <parent-ccxml-session-id>—The session was created by a <createccxml> tag from a parent session.NEWCALL <call-params>—The session was created because of an inbound call, where <call-params> records relevant parameters (for example, the UUID on the connection).

• The reason the CCXML session ended:EXIT—The <exit> tag was executed.KILL—The session was terminated by a ccxml.kill.unconditional event or an unhandled ccxml.kill event.DOCINIT—The session ended because an error was encountered during document initialization.ERROR—The session was ended by an unhandled error event.SYSERR—The session aborted because of an internal error.

• An ID indicating the source of the session:For calls started by a connection, the connection ID of the initiating call.For externally created calls, the eventsource URI.For forked sessions, the Component ID of the parent session.



OR ServiceThe OR interface enables the Resource Manager and Media Control Platform components to accumulate statistics about call arrivals and call peaks, and submit them to the Reporting Server, through the Reporting Client.• Call arrivals—Counts are derived from the CDRs as they are submitted or

updated.• Call peaks—Statistics are derived from counts of the maximum number of

concurrent calls that are observed within a given five-minute time period. The Resource Manager submits peaks for the deployment as a whole and for each VoiceXML application processed.The Media Control Platform submits peaks for itself alone.

The Reporting Client submits OR data to the Reporting Server at a configurable interval (see the ems.ors.reportinginterval parameter). The default is every minute.

Reporting ClientThe Reporting Client on each component provides reliable delivery of DATAC logs and metrics, CDRs, and OR statistics to the Reporting Server.The Reporting Client persistently queues data when the Reporting Server is unavailable, and uses exponential back-off to attempt to reconnect to the Reporting Server. Data that has been submitted to the Reporting Client will eventually be sent to the Reporting Server, even in the face of extended Reporting Server outages.

Note: Data in memory is lost if the call-processing component shuts down unexpectedly. Data is persisted to disk only if the Reporting Client is unable to successfully deliver the data to an available Reporting Server.

The Reporting Client can be configured to send CDRs and metrics in batches, to improve performance. However, this can result in slight delays in data delivery. For more information, see the description of the ems.rc.cdr.batch_size parameter on page 108.

Reporting ServerAs shown in Figure 3 on page 62, the services that the Reporting Server provides include the following:• Storage services—Logs and metrics that are delivered by the Reporting

Client on the components are stored in the GVP reporting database, where they can be queried by Reporting Web Services.

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• Reporting Web Services—HTTP web services return XML that conforms to well-defined schemas. XML-based reports display on the Monitoring tab in the Genesys Administrator. For more information, see “Reporting Web Services” on page 69.

• VAR Stats Generator—The Reporting Server computes VAR statistics based on the VAR-specific metrics it receives (see “VAR Metrics” on page 65).

• Summarization process—Every hour (on the half-hour), the Reporting Server rolls five-minute statistics into higher-level hourly, daily, weekly, and monthly summaries. The process summarizes VAR and OR data only.For performance reasons, the process does not start summarizing for a period until that period has ended. For example, a monthly summary for January will not be created until the start of February.The Reporting Server can derive summaries on the fly. (For example, you can request a monthly report for January before January has completed.) However, this puts more load on the database than when the regular summarization process derives summaries from precomputed data.

• Database Maintenance Process (DBMP)—The DBMP purges old data in accordance with data retention policies. By default, the process runs once daily, at a configurable time. The data retention policies are also configurable. For more information, see “Configuring Database Retention Policies” on page 206.

Reporting Web ServicesReporting Web Services can be deployed over HTTP or HTTPS. By default, Reporting Web Services are deployed at the following URL:http://<Reporting Server host name>:8080/ems-rs

The reporting services return results (reports) as XML documents that conform to available RelaxNG schemas. Therefore, the EMS Reporting data is available to third-party reporting products as well as through the Monitoring tab in the Genesys Administrator GUI. For detailed information about the XML schemas for GVP Reporting web services, contact Genesys Technical Support.GVP can leverage Genesys Management Framework access control mechanisms to control access to reporting data. If this feature is enabled, users must provide credentials (login ID and password) with reporting service requests, and the credentials are validated with Management Framework to determine if the user is authorized to access the reporting data. For information about configuring access control for GVP Reporting, see “Controlling Access to Reporting Services” on page 210.

Report Categories Reporting services are grouped into the following categories:• Real-time• Historical


Chapter 2: How GVP Works SNMP Monitoring

• VARFor more information about the GVP reports that are available in these categories, see the Monitoring part of this guide, starting on page 217.

SNMP MonitoringGVP 8.0 supports Simple Network Management Protocol (SNMP) monitoring for the Resource Manager, Media Control Platform, Call Control Platform, and Fetching Module components. Using SNMP in a GVP deployment is optional.To use SNMP in your GVP deployment, you must first install the Genesys SNMP Master Agent on each GVP host, as described in the Genesys Voice Platform 8.0 Deployment Guide.Each GVP Application acts as an AgentX subagent, and connects to the Genesys SNMP Master Agent on the host. Multiple subagents can register to the same master agent. The Master Agent handles all queries from the Management Data GUI and any Network Management Station (NMS), and sends AgentX queries to the respective subagent (in other words, the GVP processes).Traps, which are generated from logs, flow from the subagent to the Master Agent, and then to trap destinations as configured on the Master Agent.The traps are defined in the GVP Management Information Bases (MIBs), which are available in their own installation package (IP). For more information about the GVP MIBs, see the Genesys Voice Platform 8.0 Troubleshooting Guide.

Secure CommunicationsGVP 8.0 supports the following protocols for secure communications:• Secure SIP (SIPS)—SIP over the Transport Layer Security (TLS) protocol,

for call control and resource management messaging between the Resource Manager and the Media Control Platform and Call Control Platform resources.GVP supports TLSv1.

• Secure HTTP (HTTPS)—HTTP over Secure Socket Layer (SSL), for fetch communications among the NGI/CCXMLI, Fetching Module, and Squid caching proxy, and the Fetching Module and web application server. The Reporting Server also supports HTTPS, for receiving and responding to authenticated report requests from the web server.GVP supports SSLv2, SSLv3, and SSLv23.

User’s Guide 71

Chapter 2: How GVP Works GVP Identifiers and SIP Headers

• Secure RTP (SRTP)—A profile of RTP that provides encryption and authentication of audio and video data in RTP streams between the Media Control Platform and the Media Gateway.SRTP encryption keys and options are exchanged in SIP INVITE and response messages, preferably using SIPS.

The GVP components ship with a generic private key and SSL certificate, and default SIP transports for TLS are configured in the component Application objects. It is therefore possible to implement basic security out of the box. However, for more stringent security, Genesys recommends that you obtain your own SSL keys and certificates.For more information about obtaining SSL keys and certificates, and configuring the GVP components to use SIPS, HTTPS, and SRTP in the GVP deployment, see “Enabling Secure Communications” on page 93.

Considerations and Usage NotesBefore you implement widespread use of HTTPS in your GVP deployment, consider the following:• Complete use of SSL will have an impact on platform performance and

capacity. Lags in fetch times and high CPU usage are normal when SSL is used, because the web server must encrypt every byte of data, and then the platform must decrypt the received data. In addition, an SSL handshake takes place between the web server and the platform before data transmission starts.

• Data fetched with HTTPS will not be cached.• Before you use HTTPS to reference grammars, ensure that your ASR

engine supports it.• Be aware that, if a VoiceXML page was fetched with https and resources

within the page (such as audio files, grammars, and scripts) are referenced with a relative URI, the full URI for the resource will also use https. If you want to use HTTP to fetch a resource from a page that was fetched with HTTPS, ensure that the VoiceXML page explicitly references the resource as an http URI.

GVP Identifiers and SIP HeadersThis section explains two important categories of identifiers that are used in GVP:• Session Identifiers• Application Identifiers (see page 72)



Session IdentifiersThere are three types of session identifiers that are used to track, co-ordinate, and report on GVP sessions. Table 5 describes the session identifiers and the SIP extension headers in which the ID information is captured.

Application IdentifiersThere are two kinds of applications in a GVP deployment:• The GVP components or processes, which exist as Application objects in

the Genesys Configuration Layer.• The VoiceXML and CCXML applications, which exist as IVR Profile

objects in the Genesys Configuration Layer.Table 6 describes the identifiers that GVP uses for both kinds of applications, and the SIP extension headers in which the ID information is captured.

Table 5: GVP Session Identifiers and SIP Headers

Session ID Description SIP Header

Genesys CallUUID The Universally Unique Identifier (UUID) that is generated by T-Server or SIP Server for the customer interaction.

X-Genesys-CallUUIDThe Resource Manager (SIP Proxy) and GVP components (User Agents) propagate this header without changes in all SIP messages.

GVP Session ID The 128-bit Globally Unique Identifier (GUID) that identifies a call session for a particular GVP resource. The GUID is generated by:Resource Manager when it creates a new session in response to a new SIP INVITE request.Media Control Platform or Call Control Platform when it initiates a new session for an outbound call.

X-Genesys-GVP-Session-IDIf the header does not already exist in the SIP request, the Resource Manager inserts the header in requests that it forwards. The Resource Manager and GVP components propagate this header in all subsequent SIP messages for the session.

GVP Component ID

The ID that is generated by the GVP component to identify the call leg.The component correlates the Component ID with the GVP Session ID, and logs the correlation for correct call detail records (CDRs).

gvp.ccp.sessionid—For the Call Control Platform only, when it sends a request to initiate an outbound session and the Resource Manager session has not been assigned yet. This allows a newly created CCXML session to make multiple SIP requests before it has received a response to any of them.

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Importing and Exporting Configuration Server Data

When data is exported from Configuration Server and then imported back with or without modification, be aware that the DBIDs of existing configuration objects (such as GVP Application processes and IVR Profiles) may change. In these cases:• Reporting Server will not be able to correlate historical data with the new

IDs, and GVP reports will not display older data.

Table 6: GVP Application Identifiers and SIP Headers

Application ID Description SIP Header

Application DBID The DBID assigned to the GVP component Application by the Configuration Layer.This ID is used internally by the Reporting Server to generate reports. You can view the ID in the Genesys Administrator on the Deployment > Deploy > Deploy GVP > <GVP Solution Deployment Scenario> > 3. Summary tab.

N/A

IVR Profile name The user-defined name that was assigned to the IVR Profile when it was created.Note: Spaces are not allowed in the IVR Profile name.A GVP component can control the voice or call control application for a new call leg session, by extracting the value of the gvp.rm.tenant-id parameter from the X-Genesys-GVP-Session-ID header of an existing session, and using this value for the gvp-tenant-id parameter in a new SIP request.

• gvp-tenant-id parameter in the SIP Request-URI—The Resource Manager uses this parameter, if present, to identify the voice or call control application for a new session. For more information, see “Mapping the Call to an IVR Profile” on page 30.

• gvp.rm.tenant-id parameter in the X-Genesys-GVP-Session-ID extension SIP header—The Resource Manager inserts this parameter in the header before it forwards the initial session request.

IVR Profile DBID The DBID assigned to the IVR Profile by the Configuration Layer.

X-Genesys-RM-Application-dbidThe Resource Manager adds this header to the initial INVITE request to a resource. Resources log the DBID in their CDRs to the Reporting Server.



• GVP components that use CCILib may encounter problems because they will no longer receive updates for objects for which they registered to receive updates under the old IDs. Restarting Configuration Server will fix this problem.

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Part

2 Provisioning GVPThis part of the manual provides information about Genesys Voice Platform (GVP) configuration and provisioning that you perform on the Provisioning tab of the Genesys Administrator.This information appears in the following chapters:• Chapter 3, “Configuration and Provisioning Overview,” on page 77• Chapter 4, “Configuring Common Features,” on page 85• Chapter 5, “Configuring the Resource Manager,” on page 123• Chapter 6, “Provisioning IVR Profiles,” on page 139• Chapter 7, “Configuring the Media Control Platform,” on page 157• Chapter 8, “Configuring the Call Control Platform,” on page 183• Chapter 9, “Configuring the Fetching Module and Squid Proxy,” on

page 197• Chapter 10, “Configuring the Reporting Server,” on page 203


User’s Guide 77

Chapter

3 Configuration and Provisioning OverviewThis chapter provides an overview of the tasks to configure Genesys Voice Platform (GVP) components and provision GVP.It contains the following sections:

Configuring GVP, page 77Task Summary: Configuring GVP, page 81

Note: This guide assumes that you have deployed a basic GVP as described in the Genesys Voice Platform 8.0 Deployment Guide. For more information about installing GVP components and providing the basic connections, see the Deployment Guide.

Configuring GVPThe GVP components are configured as Application objects in the Genesys Configuration Layer. To deploy the Voice Platform Solution (VPS), you must create and configure the required Application objects in the Genesys Administrator. For information about creating and deploying the GVP Applications, see the Genesys Voice Platform 8.0 Deployment Guide.To make the voice and call control applications available for GVP to use, you must provision the IVR Profiles in the Genesys Administrator. To trigger the launching of a particular VoiceXML or CCXML application when an incoming call is received, map the IVR Profile to a DNIS range. For more information about provisioning IVR Profiles and, if required, mapping them to DNs, see Chapter 6 on page 139.


Chapter 3: Configuration and Provisioning Overview Configuring GVP

Configuring GVP Processes in the Genesys AdministratorThe following procedure describes how to configure GVP Application and IVR Profile objects in the Genesys Administrator. For more information about using the Genesys Administrator, see the online Framework 8.0 Genesys Administrator Help.

Procedure:Viewing or modifying GVP configuration parameters

Purpose: To describe the general method to use the Genesys Administrator to view or modify configuration options in GVP Application and IVR Profile objects.

Prerequisites

• The Application or IVR Profile object has been created as described in the Genesys Voice Platform 8.0 Deployment Guide. In particular, for GVP Application objects, the Application was created from an Application Template into which metadata had been imported.

• You are logged in to the Genesys Administrator. To access the Genesys Administrator, go to the following URL:http://<Genesys Administrator host>/wcm

Start of procedure

1. In the Genesys Administrator, go to the Settings tab of the object you want to configure:• For a component Application, go to the Provisioning > Environment

> Applications > <Component Application> > Settings tab.• For an IVR Profile, go to the Provisioning > Voice Platform > IVR

Profile > <IVR Profile Property> > Settings tab.Figure 4 and Figure 5 on page 80 show the Settings tab.

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Figure 4: Settings Tab

For each configurable parameter, the Settings tab displays the following information:• A plain-language display name.• The configuration section that contains the option.• The configuration option name, as it would appear on the Options tab.• The current option value, either user-defined or default. User-defined

values display in bold.User-defined values also appear on the Options tab.

2. You can change the display in a number of ways:• To sort the information in ascending or descending order by column,

click the column header to activate a drop-down list, and select the desired sort order option.

• To show or hide a column, click any column header to activate a drop-down list, select the Columns submenu, and select or clear check boxes in the Columns list to show or hide columns.

Display Name Section Option Name Option Value Display FilterReload



• To filter the options that are displayed, select a different grouping from the Display Filter drop-down list.For example, the Media Control Platform Application provides the following option groupings:• Media Control Platform Main Settings• Logging• GVP Logging and Reporting• RemoteDial Application Module Settings• Conference Application Module Settings• Media Processing Settings• Session Manager Settings• SIP Settings• Speech Resource Management Settings• SNMP Settings• NGI Settings• All Settings

• To view more information about an option, click the arrow to the left of the display name. A description of the option appears (see Figure 5).

Figure 5: default Application Settings Tab, GVP Reporting Filter

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Chapter 3: Configuration and Provisioning Overview Task Summary: Configuring GVP

3. To change an option setting:a. Click the Name or Value of the option that you want to change.

A dialog box appears, which displays a full description of the option, its current value, valid values, default value (if applicable), and when changes takes effect.You can view the same information for all configuration options in the Genesys Voice Platform 8.0 Configuration Options Reference.

b. Enter the new value in the Value field or, if applicable, select the option from the Value drop-down list. In cases where multiple values can be selected, hold down Ctrl to select multiple values.

c. Click OK.4. To save your changes, click Save and then Apply.

Option values that you change on the Settings tab get updated on the Options tab as well.Similarly, options that you change on the Options tab get updated on the Settings tab. However, be aware that the Settings tab displays options and option names only as defined in the component metadata.

5. To update the metadata descriptions (from an updated Templates XML file), click Reload. This reloads the metadata file without affecting configured option values.

End of procedure

Task Summary: Configuring GVPTable 7 provides an overview of the tasks to implement full GVP functionality in your deployment.



Table 7: Configuring and Provisioning GVP

Objective Related Procedures and Actions

Set up connections, SIP communications, and routing between the Resource Manager and all the other GVP components.

See:• “Configuring SIP Communications and Routing” on page 86.• For secure communications, see “Enabling Secure

Communications” on page 93.• For High Availability, see “Enabling High Availability” on

page 134.See also component-specific requirements:• For the Media Control Platform, see Table 28 on page 158.• For the Call Control Platform, see Table 31 on page 184.

Provision the resources for the Resource Manager.

See “Configuring Logical Resource Groups” on page 126.

Provision the IVR Profiles. See Chapter 6 on page 139.

Enable GVP Reporting. • Configure the options in the ems configuration section of the Resource Manager, Media Control Platform, Call Control Platform, Fetching Module, and, if applicable, Cluster Manager Application objects. For more information, see “Configuring EMS Reporting” on page 103.

• Configure the Reporting Server (see Chapter 10 on page 203).• If required, configure access control for Reporting services

(see “Controlling Access to Reporting Services” on page 210).• On the Monitoring tab of the Genesys Administrator, verify

that the Voice Platform view appears in the navigation panel. If necessary, modify the default (Configuration Manager) Application configuration to enable GVP reports to display in the Genesys Administrator. For more information, see “Connecting to the Genesys Administrator” on page 213.

(Optional) Enable Automatic Speech Recognition (ASR) and Text-to-Speech (TTS).

See “Enabling ASR and TTS” on page 159.

User’s Guide 83


(Optional) Enable conferencing. See “Enabling Conference Services” on page 102.

(Optional) Configure individual components to customize or enable GVP features.

In general, see the remaining chapters in the Provisioning part of this guide. More specifically, to customize:• Logging behavior, see “Configuring Logging” on page 111.• Session behavior and performance, see “Configuring Session

Timers and Timeouts” on page 117.• Messaging, see “Customizing SIP Responses” on page 116

and Table 64 on page 279.• Call Control Platform device profiles, see “Configuring

Device Profiles” on page 188.• Caching behavior, see “Configuring the Squid Caching

Proxy” on page 200.

Table 7: Configuring and Provisioning GVP (Continued)




User’s Guide 85

Chapter

4 Configuring Common FeaturesThis chapter describes how to implement functionality that is shared across all the components in a Genesys Voice Platform (GVP) deployment.It contains the following sections:

Configuring SIP Communications and Routing, page 86Enabling Secure Communications, page 93Enabling Conference Services, page 102Configuring EMS Reporting, page 103Configuring Logging, page 111Customizing SIP Responses, page 116Configuring Session Timers and Timeouts, page 117

This chapter describes selected configuration options (parameters) that are common to GVP components. Later chapters similarly highlight important configuration options that are more component-specific.The configuration options tables provide parameter descriptions as well as the default parameter values that are preconfigured in the GVP Application objects. For information about all the available configuration parameters, see the Genesys Voice Platform 8.0 Configuration Options Reference.Configure the options in the Genesys Administrator on the Provisioning >

Environment > Applications > <GVP Component> > Settings tab. For the detailed steps to configure option settings, see Viewing or modifying GVP configuration parameters, page 78.


Chapter 4: Configuring Common Features Configuring SIP Communications and Routing

Note: The configuration parameters for GVP processes are complex and provide a great deal of flexibility. Deploying GVP as described in the Genesys Voice Platform 8.0 Deployment Guide provides a fully functional, basic GVP deployment, with the minimum customizations required for GVP to operate in your environment. Ensure that you review the configuration options and fully understand their implications before you perform additional customizations.

Configuring SIP Communications and Routing

Table 8 summarizes the steps and parameters to configure the transport and routing mechanisms for SIP messaging within the GVP deployment.

Table 8: Task Summary: Configuring SIP Communications and Routing


For each Resource Manager, Media Control Platform, and Call Control Platform Application in your deployment, configure the SIP transports for the supported transport protocols.

Configure the sip.transport.<x> options (see page 92). Note the following:• For the Resource Manager, specify separate transports for SIP

proxy, registrar, and monitoring purposes.• The lowest <x> in a set of sip.transport.<x> options

indicates the preferred default protocol. By default, UDP is the preferred protocol for all components (sip.transport.0).

• To make TCP the preferred protocol, either rename (in other words, reorder) the respective sip.transport parameters, or else remove the default sip.transport.0 UDP parameter so that the default sip.transport.1 TCP parameter is the lowest defined sip.transport.<x>.

• For secure SIP (SIPS) communications, specify a transport for TLS. For more information, see “Enabling Secure Communications” on page 93.

Note: If you change the default preferred protocol for the Call Control Platform, there are additional steps you must take on the CCXML application side to ensure that the Request-URI specifies the correct protocol. For more information, see page 184.

User’s Guide 87


Table 9 provides information about important SIP communications and routing options. Table 9 provides parameter descriptions as well as the default parameter values that are preconfigured in various configuration sections in the Resource Manager, Media Control Platform, and Call Control Platform Application objects.For information about all the available configuration options, see the Genesys Voice Platform 8.0 Configuration Options Reference.

Configure the route set and routing table for outbound calls.

• For each Media Control Platform and Call Control Platform Application in your deployment, configure the sip.routeset or sip.securerouteset option (see page 90 or 91).

• For each Resource Manager, Media Control Platform, and Call Control Platform Application in your deployment, configure the required sip.route.dest.<n> entries (see page 89).

Verify settings that determine behavior in relation to the SIP stack.

Review and, if necessary, modify the options that control such parameters as number of threads, size of the Maximum Transmission Unit (MTU) of the network interfaces, and number of connections.• For the Resource Manager, relevant options are in the proxy

configuration section.• For the Media Control Platform and Call Control Platform,

relevant options are in the sip configuration section.

Table 8: Task Summary: Configuring SIP Communications and Routing (Continued)




Table 9: Selected SIP Communications and Routing Options

Section.Option Name

Description Valid Values and Syntax

Resource Manager:• proxy.

sip.route.default.<protocol>

Media Control Platform and Call Control Platform:• sip.

route.default.<protocol>

The default route for messages for the <protocol>, where <protocol> is:• tcp—The parameter defines the default route

for Transmission Control Protocol (TCP) messages.

• tls—The parameter defines the default route for Transport Layer Security (TLS) messages.

• udp—The parameter defines the default route for User Datagram Protocol (UDP) messages.

The value of this parameter is a number that corresponds to the transport interface index (<x>) for a transport defined by sip.transport.<x> (see page 92).The default transport is used when the component application finds no routes for the <protocol> in the routing table (see sip.route.dest.<n> on page 89). If the sip.route.default.<protocol> parameter is not set, the first UDP transport found in sip.transport.<x> becomes the default.

Any unsigned integer.Default value: Empty

User’s Guide 89



sip.route.dest.<n>Media Control Platform and Call Control Platform:• sip.

route.dest.<n>

Each <n> represents an entry in a routing table. The value of this parameter specifies the content of the entry in the routing table. The component application searches the routing table to identify the SIP transport and network interface to use for outbound calls.The order of the sip.route.dest.<n> parameters matters, because the component application searches the routing table linearly until it finds a <destination> entry that matches the requested, masked IP address.If the masked IP address matches a destination entry, that route is accepted. The <transport> part of the routing table entry then determines the transport to use.• In most cases, the first accepted route will be

used.• If the protocol is specified or required (for

example, TCP must be used when the message size is larger than sip.mtusize), the accepted route must also match the required protocol. If there is no routing table entry with a transport that matches the specified protocol, the default route of that protocol will be used.

• If no route is found, the default route for the protocol in sip.transport.0 will be used.

Example:sip.route.dest.0=138.120.72.0 255.255.255.0 1 0

Say a call is made to a host at 138.120.72.20. Applying the <netmask> of 255.255.255.0 to the outgoing IP address yields 138.120.72.0, which matches the defined <destination> in the route. Therefore, the transport in sip.transport.1 will be used.Note: In the Resource Manager configuration, there must be a proxy.sip.route.dest.<n> parameter defined for each <n> in the parameter proxy.sip.route.dests (see page 90).

A string representing <destination> <netmask> <transport> <metric>

where:• <destination> is the

subnet IP address of the network interface.

• <netmask> is the network mask that the component application applies in evaluating whether the requested IP address matches the <destination> entry.

• <transport> corresponds to the transport interface index (<x>) for the applicable transport, as defined by sip.transport <x> (see page 92).

• <metric> is the routing metric. (Not used in GVP 8.0)

Default value: Empty

Table 9: Selected SIP Communications and Routing Options (Continued)

Section.Option Name




(For Resource Manager only)proxy.sip.route.dests

An index list for entries in a routing table. A space-delimited list of integers, starting from 0 and incrementing by 1 for each entry.Example:0 1 2 3 specifies four entries in the routing table.Default value: Empty

(For Media Control Platform and Call Control Platform only)sip.routeset

A comma-separated list of SIP Proxy addresses that defines a route set for non-secure SIP outbound calls. If defined, this route set is inserted as the ROUTE header for all outgoing calls. This forces GVP to use the defined route set for SIP messages.Using the lr parameter with the URI (see syntax) forces the User Agent Client (UAC) to place the remote target URI into the Request-URI and to include the route set in the ROUTE header.The SIP port on SIP Proxies is usually 5060.Example: <sip:RM_host.yourdomain.com:5060;lr>,<sip:Proxy2.yourdomain.com:5060;lr>—The Media Control Platform or Call Control Platform will send the outgoing request to the Resource Manager, which will, in turn, route the request to Proxy 2, which will redirect the message to its intended destination.Note: The route set does not apply to SIP REGISTER messages.

<sip:<Resource Manager IP address>:<Resource Manager SIP port>;lr>[,<sip:<Next SIP Proxy or UA IP address>:<Proxy SIP port>;[lr]>,...]

Note: The outer angle brackets are required characters in the string.Default value: Empty


Section.Option Name


User’s Guide 91


(For Media Control Platform and Call Control Platform only)sip.securerouteset

A comma-separated list of SIP Proxy addresses that defines a route set for secure SIP outbound calls. If defined, this route set is inserted as the ROUTE header for all outgoing calls. This forces GVP to use the defined route set for secure SIP messages.Using the lr parameter with the URI (see syntax) forces the UAC to place the remote target URI into the Request-URI and to include the route set in the ROUTE header.The secure SIP port on SIP Proxies is usually 5061.

Note: The route set does not apply to SIP REGISTER messages.

• <sips:<Resource Manager IP address>:<Resource Manager secure SIP port>;lr>[,<sips:<Next SIP Proxy or UA IP address>:<Proxy SIP port>;[lr]>,...]

• <sip:<Resource Manager IP address>:<Resource Manager secure SIP port>;transport=tls;lr>[,<sip:<Next SIP Proxy or UA IP address>:<Proxy SIP port>;transport=tls;[lr]>,...]

Note: The outer angle brackets are required characters in the string.Default value: Empty


Section.Option Name




Table 10 summarizes the default values for the sip.transport.<n> parameter for the Resource Manager, Media Control Platform, and Call Control Platform.


sip.transport.<x>• registrar.

sip.transport.x• monitor.

sip.transport.xMedia Control Platform and Call Control Platform:• sip.

transport.<x>

The parameters that define the transport layer for SIP stack and the network interfaces that are used to process SIP requests.<x> is the transport interface index that identifies the transport, so that you can specify different combinations of parameters for different protocols.The Resource Manager specifies separate SIP transports for messages relating to its role as SIP Proxy (INVITE messages), SIP Registrar (REGISTER messages), and monitor (OPTIONS messages).The default transport is the smallest non-empty transport interface index (transport.<x>).For Secure SIP, ensure that you specify Transport Layer Security (TLS) as the transport layer protocol (<transport_type> = tls), and include the following additional SIP transport parameters:• cert=<TLS certificate path and file name>

(required)• key=<TLS key path and file name> (required)• type=<type of secure transport> (optional)

Valid values: TLSv1|SSLv2|SSLv3|SSLv23 Default value: SSLv23

• password=<password associated with the certificate and key pair> (required only if the key file is password protected)

Note: For each transport that you define with a specific IP address (in other words, the value of <ip> is not any), you must also configure at least one entry in the routing table (see sip.route.dest.<n> on page 89), or the interface will never be used.

<transport_name> <transport_type>:<ip>:<port> [<parameters>]

where:• <transport_name> is any

alphanumeric string.• <transport_type> is the

transport layer protocol: udp|tcp|tls.

• <ip> is the IP address of the network interface that accepts incoming SIP messages (the default value of any means all network interfaces).

• <port> is the port number where SIP stack accepts incoming SIP messages.

• [<parameters>] are any additional, optional SIP transport parameters.

Default values: See Table 10 on page 93.


Section.Option Name


User’s Guide 93

Chapter 4: Configuring Common Features Enabling Secure Communications

Enabling Secure CommunicationsTable 11 summarizes the steps and parameters to set up your GVP deployment to use Secure Socket Layer (SSL) technology for secure SIP (SIPS), secure HTTP (HTTPS), and secure RTP (SRTP) communications.

Table 10: Default SIP Transports

Component Application Section.Option Name Default Value

Resource Manager proxy.sip.transport.0 transport0 udp:any:5060

proxy.sip.transport.1 transport1 tcp:any:5060

proxy.sip.transport.2 transport2 tls:any:5061 cert=$InstallationRoot$\config\x509_certificate.pem key=$InstallationRoot$\config\x509_private_key.pem

registrar.sip.transport.0 transport0 udp:any:5062

registrar.sip.transport.1 transport1 tcp:any:5062

registrar.sip.transport.2 transport2 tls:any:5063 cert=$InstallationRoot$\config\x509_certificate.pem key=$InstallationRoot$\config\x509_private_key.pem

monitor.sip.transport.0 transport0 udp:any:5064

monitor.sip.transport.1 transport1 tcp:any:5064

monitor.sip.transport.2 transport2 tls:any:5065 cert=$InstallationRoot$\config\x509_certificate.pem key=$InstallationRoot$\config\x509_private_key.pem

Media Control PlatformNote: If all sip.transport.x values are empty, UDP, TCP, and TLS transports will all be enabled and respectively listen from ports 5060, 5060, and 5061 on any network interface.

sip.transport.0 transport0 udp:any:5070

sip.transport.1 transport1 tcp:any:5070

sip.transport.2 transport2 tls:any:5071 cert=$InstallationRoot$\config\x509_certificate.pem key=$InstallationRoot$\config\x509_private_key.pem

Call Control Platform sip.transport.0 transport0 udp:any:5068



For general information about secure communications in GVP, see “Secure Communications” on page 70.

Note: The GVP components support SIPS, but the Genesys SIP Server does not. Before you enable SIPS in your GVP deployment, contact your Genesys Sales Representative for more information.

Table 11: Task Summary: Enabling SIPS, HTTPS, and SRTP in GVP


Set up GVP to use SIPS for call control messaging.

1. If required, generate and deploy the SSL private key and certificate (see Creating an SSL private key and certificate).

2. On the Resource Manager, Media Control Platform, and Call Control Platform Applications, specify the SIP transport for TLS, including the additional parameters for the certificate and key (see information about the sip.transport.<x> option on pages 86 and 92, and the default values in Table 10 on page 93).

3. On the Media Control Platform and Call Control Platform Applications, specify secure routing for outbound calls (see information about the sip.securerouteset option on page 91 and the sip.route.dest.<n> option on page 89).

4. Modify the CCXML applications, as required, to ensure that the Request-URI specifies TLS as the transport protocol. For more information, see page 184.

Set up the Fetching Module to use HTTPS.

1. Generate and deploy the SSL private key and certificate. For information about creating a self-signed certificate, see Creating an SSL key and self-signed certificate for use with IIS, page 97.

2. On each Fetching Module Application in your deployment, configure the Fetching Module process (pwproxy) to access files over HTTPS (see Configuring the Fetching Module for HTTPS, page 99).

Configure additional iproxy.ssl_* options as required for your deployment.

3. Modify the Squid configuration file, if necessary, to configure “safe” and SSL ports, and to enforce SSL (see Modifying the Squid Configuration, page 200).

User’s Guide 95


Verify that timeout settings are suitable for your deployment.

Given the additional processing time and lags associated with SSL encryption/decryption and handshakes, reconsider the following settings in particular:• For the Fetching Module, iproxy.connect_timeout (default is 5 seconds).

• For the Media Control Platform, timeouts in the sessmgr and sip sections.

Enable SRTP for the media channel between the Media Control Platform and the remote endpoint.

On the Media Control Platform Application:1. Specify the required mode (accept-only or offer) in the

mpc.srtp.mode parameter. By default, SRTP is not enabled.2. If necessary, modify the default values for the encryption

and authentication algorithms (the cryptographic suites) and session parameters that the Media Control Platform will advertise in the SDP crypto attribute:mpc.srtp.cryptomethodsmpc.sessionparamsmpc.sessionparamsoffer

Enable SRTP for the media channel between the MRCPv2 server and the Media Control Platform.

On the MRCPv2 Application that represents the third-party MRCP server for ASR or TTS, verify and, if required, modify settings for the following options:provision.vrm.client.TlsCertificateKeyprovision.vrm.client.TlsPrivateKeyprovision.vrm.client.TlsPassword

If necessary, set up the Reporting Server web server to use HTTPS.

• If the Reporting Server Tomcat is deployed primarily as a Servlet/JSP container behind another web server (such as Apache or Microsoft IIS), no further configuration is required.

• If the Reporting Server Tomcat is deployed as a stand-alone web server, you must generate and deploy the SSL private key and certificate, and then modify the Tomcat configuration to use native SSL. For more information, see Configuring a stand-alone Tomcat web server for SSL, page 101.

See also “Enabling HTTP Basic Authorization for Reporting” on page 211.

Configure Genesys Administrator to use HTTPS to access Reporting Server web services, for the GVP reports that display in the Monitoring > Voice Platform view.

• In the Genesys Administrator, on the Provisioning > Environment > Applications > default > Settings tab, set the value of rptui.enablehttps to true.

• Ensure that the web server for Reporting Server is configured to enable HTTPS.

Table 11: Task Summary: Enabling SIPS, HTTPS, and SRTP in GVP (Continued)




Note: Observe standard security practices to ensure that you protect the security of SSL private keys, SSL certificates, and configured user names and passwords—for example, ensure that they are stored on secure hosts, and do not create them over a network.

The following procedures support the tasks outlined in Table 11:• Creating an SSL private key and certificate• Creating an SSL key and self-signed certificate for use with IIS, page 97• Configuring the Fetching Module for HTTPS, page 99

Procedure:Creating an SSL private key and certificate

Purpose: To provide an example of a way to create and deploy the private key and SSL certificate that are used for SIPS and HTTPS authentication.

Perform this procedure for each Resource Manager, Media Control Platform, and Call Control Platform in your deployment.

Prerequisites

• The OpenSSL Toolkit (openssl) or other SSL tool is available.You can download the OpenSSL Toolkit for Windows from Shining Light Productions at the following URL: http://www.shininglightpro.com/products/Win32OpenSSL.html

For more information about OpenSSL, see http://www.openssl.org/.

Start of procedure

1. Generate the private key.• For a password-protected key, execute the following command:

openssl genrsa -aes128 -out x509_private_key.pem 2048

• For a non-password-protected key, execute the following command:openssl genrsa -out x509_private_key.pem 2048

2. Generate the certificate.The following example of the required command creates a certificate with file name x509_certificate.pem, which expires in 1095 days:openssl req -new -x509 -key x509_private_key.pem -out x509_certificate.pem -days 1095

For information about additional supported parameters, see the openssl Manual page on the OpenSSL web site (http://www.openssl.org/).

User’s Guide 97


3. Install the certificate and key.The default GVP configuration assumes that the file names and paths are as follows:• For the certificate:

$InstallationRoot$\config\x509_certificate.pem

• For the private key:$InstallationRoot$\config\x509_private_key.pem

End of procedure

Next Steps

• If required, modify the sip.transport.<x> configuration option for TLS to update the parameters for the certificate path, key path, and password (if applicable).

Procedure:Creating an SSL key and self-signed certificate for use with IIS

Purpose: To provide an example of a way to use the OpenSSL Toolkit to create a private key and self-signed SSL certificate request, to enable HTTPS connections to the IIS web server for Fetching Module communications.

Prerequisites

• The OpenSSL Toolkit (openssl) has been installed, with default settings.You can download the OpenSSL Toolkit for Windows from Shining Light Productions at the following URL: http://www.shininglightpro.com/products/Win32OpenSSL.html

For more information about OpenSSL, see http://www.openssl.org/.

Start of procedure

1. Set up the openssl directories and files.a. (Optional, but recommended) Add C:\OpenSSL\bin to your system path

(Control Panel > System > Advanced > Environment Variables > System Variables).

b. Create a working directory—for example, C:\ssl.c. Create the directory structure and files required by openssl:

• Directories: keys, requests, and certs.



• Files: database.txt and serial.txt—these are empty (zero-byte) text files.

To create the directories and files manually, execute the following commands at the C:\ssl> UNIX prompt:md keys

md requests

md certs

copy con database.txt

^Z

copy con serial.txt

01

^Z

2. Set up a Certificate Authority (CA).a. At the C:\ssl> prompt, execute the following command to create a

1024-bit private key:openssl genrsa -des3 -out keys/ca.key 1024

b. At the C:\ssl> prompt, execute the following command to create the CA certificate:openssl req -config openssl.conf -new -x509 -days 1001 -key

keys/ca.key -out certs/ca.cer

The following certificate is created:c:\ssl\certs\ca.cer

3. Create an IIS Certificate Request (certreq.txt).For more information, see the Microsoft Knowledge Base article number 228821, which is available from Microsoft Technical Support (http://support.microsoft.com).

4. Sign the Certificate Request.a. Copy the certreq.txt file into C:\ssl\requests.b. At the C:\ssl> prompt, execute the following command to sign the

request:C:\ssl>openssl ca -policy policy_anything -config openssl.conf

-cert certs/ca.cer -in requests/certreq.txt -keyfile keys/ca.key

-days 360 -out certs/iis.cer

5. Install the new certificate under IIS.For more information, see the Microsoft Knowledge Base article number 228836, which is available from Microsoft Technical Support (http://support.microsoft.com).The secure web server is now accessible from any web browser, using SSL.

End of procedure

User’s Guide 99


Next Steps

• Configure the Fetching Module (pwproxy) to access files over HTTPS (see Configuring the Fetching Module for HTTPS).

Procedure:Configuring the Fetching Module for HTTPS

Purpose: To modify the default Fetching Module configuration to enable secure HTTPS communications.

Perform this procedure on each Fetching Module in your deployment.

Prerequisites

• The SSL certificate and key have been created and installed under IIS. For information about creating a self-signed certificate, see Creating an SSL key and self-signed certificate for use with IIS, page 97.

Start of procedure

1. Create the PEM certificate file.a. Using a text editor, create a new file, proxy_client.pem. You can store

the file under any directory on the Fetching Module server.b. Open the ca.key file created by openssl (see Step 2 on page 98).c. Copy all the lines from ca.key into the new proxy_client.pem file.d. Press Enter to create one blank line at the end of the text.e. Open the iis.cer file created when you signed the certificate request

(see Step 4 on page 98).f. Copy all the lines from iis.cer starting from -----BEGIN

CERTIFICATE---- and ending with -----END CERTIFICATE-----, inclusive, into the bottom of the proxy_client.pem file.

The final file will look similar to Figure 6.



Figure 6: Sample PEM Certificate File

2. Save the file.3. In the Genesys Administrator, on the Provisioning > Environment >

Applications > <Fetching Module> > Settings tab, modify the Fetching Module configuration:a. Verify that the value of the iproxy.https_proxy parameter is empty

(disabled—encrypted pages will not be cached).b. Configure the following options in the iproxy configuration section:

• ssl_key_passwd = <Your private key passphrase>• ssl_key = <Local path to your proxy_client.pem file>

-----BEGIN RSA PRIVATE KEY-----

Proc-Type: 4,ENCRYPTED

DEK-Info: DES-EDE3-CBC,70D8F72D9BB079C3

aFnOkM5agLiG7gvcEBjZ+GIAKfFsCQKuq3cYBkng/Zlp5vgapDqx6JUycPcBs7A/

Y35h4E4HDJv40gJ3xqLc4ENrhFH4Vezc4hFDb5SfQteVQP1nklxYBE5vUY+55xwv

UCcbrpD3PjqVakWPwdz7HtA7prH/4izUytE99yEE3C5pf3QpnUvOps9OH+WN3x9L

IAWun2t2bojDjwofIREx4C0iWH/3PHi9gqpbZeRXvgwvEfw8dpKwh/oV5mCexcWt

YTJ/6Nf5fFCA2NxoaboZXIBa83ISOuceZXAb5yEiXfpe4k4wPweLHc7kzhwLiwJL

6JUnG7yjAcVxeN6gDk+oxGRkPoz7xpOVwTWRk/uCSF0umai30Mrv8CuOdya0hB/2

jBD1PeH8+1yfngH5RcU33vZJIMJtHVBiTA330YQLDqke2xvJf4uBxdawU7BSmYpT

Bo35suRc4wARf7TF8Gvxl5epFDCSx32i81rkbZhv9GlFfajiiBV3VRTMlN+ydSXb

QnLU+Oe5ln1BRbY7OUXOHLuGJRMdY1j/vkJYPbCeGhOa4S4wPQT1tPYcBpYdVhcH

DFZn55Glzlf0d4BUXeFl1lKu5FK9P0B4ozLtXwMZtaUXQ44vLjPJTWLMLpNY3AKS

zmb2boDqn5btipuxwmqXYFLfIZl6h32sLLuZex3gv9lbURsD8Zr+HgqVNzXwJTW9

kDEndj5Bd+pMUe3i/9gr0nPAVMFkFsUuHEZPNNNl2AZsWwOkPsd9o7YEHVJGovS6

AV3D6KPhOHFhg8AHmrEHcJLkN77JTSlbUJdAO+t/KNyYRs3TLwEexg==

-----END RSA PRIVATE KEY-----

-----BEGIN CERTIFICATE-----

MIICYjCCAcugAwIBAgIBATANBgkqhkiG9w0BAQQFADCBijELMAkGA1UEBhMCQ0Ex

EDAOBgNVBAgTB09udGFyaW8xEDAOBgNVBAcTB01hcmtoYW0xEDAOBgNVBAoTB0dl

bmVzeXMxCjAIBgNVBAsTATExFzAVBgNVBAMTDjEzOC4xMjAuODQuMTQ0MSAwHgYJ

KoZIhvcNAQkBFhFxYUB2b2ljZWdlbmllLmNvbTAeFw0wODA0MjUxNDUxNTBaFw0w

OTA0MjAxNDUxNTBaMGMxCzAJBgNVBAYTAkNBMQswCQYDVQQIEwJPTjEQMA4GA1UE

BxMHTWFya2hhbTEQMA4GA1UEChMHR2VuZXN5czEKMAgGA1UECxMBMTEXMBUGA1UE

AxMOMTM4LjEyMC44NC4xNDQwgZ8wDQYJKoZIhvcNAQEBBQADgY0AMIGJAoGBAMQr

f+mPgVE8Aemgbg90UocmyEJ0lh2yC3KGjC1UgLPF6TQaJ2vLIniCBSUUINgDrujo

8PKMJBXOpL4FUIqMzcX0foVCx4zXK7bPwO8mibm1DB1DDJ7dy+2n7vLRV1PM6R/r

G4l2oYeRC6tYLEg3818WJnGR49yrWPMGfHvXeHTzAgMBAAEwDQYJKoZIhvcNAQEE

BQADgYEAaVG00qOsU7L3riGomIWCnp30rtcv4lnVmUDG9BvhBWNF65EfXiSwEjqi

GIAnss9CZYw1odqo+hZsNLttwERlRn973K4GGmywQFnErei5hZeonMFm0BZjkvch

ynbPVTr/O0Ot3+cKhW1Ef1osh5fFxlWlhNrwv11mpkG0OZ8pVME=

-----END CERTIFICATE-----

User’s Guide 101


• ssl_cipher_list = TLSv1For information about other SSL-related configuration options for the Fetching Module, see the Genesys Voice Platform 8.0 Configuration Options Reference (options beginning with ssl_ in the iproxy section).

c. Click Save or Apply to save the configuration changes.4. Restart the Fetching Module application.

End of procedure

Next Steps

• If required, modify the Squid configuration file to identify the “safe” ports for HTTP and SSL requests, to identify the ports to be used for SSL connections, and to deny access to non-SSL connections.For more information, see Modifying the Squid Configuration, page 200.

Procedure:Configuring a stand-alone Tomcat web server for SSL

Purpose: To provide an example of a way to configure SSL support on Tomcat, when Tomcat is deployed as a stand-alone web server for Reporting Server.

Start of procedure

1. Generate the private key and certificate for Tomcat.• To generate the certificate with openssl, execute the following

command:openssl pkcs12 -export -infile mycert.crt -inkey mykey.key \

-outfile mycert.p12 -name tomcat -CAfile myCA.crt \-caname root -chain

For information about obtaining the OpenSSL Toolkit (openssl), see the Prerequisites item on page 97.

• To generate a self-signed certificate with the Java keytool, execute the following command:$JAVA_HOME/bin/keytool -genkey -alias tomcat -keyalg RSA

2. Configure Tomcat to use native SSL.For example, add the following element in the $CATALINA_HOME/conf/server.xml file:<-- Define a SSL Coyote HTTP/1.1 Connector on port 8443 -->

End of procedure

Enabling Conference ServicesTable 12 summarizes the steps and parameters to configure the GVP deployment to provide conference service.

Note: Values for options such as conference reserve, maximums for number of conferences and participants, and conference capabilities can be set at the level of the resource group, the resource, and the IVR Profile, in order of override priority. These parameters are significant in determining how the Resource Manager handles a particular request for conference service (see, for example, “Selecting a Resource for Conference Services” on page 34). Genesys recommends that, before you modify options, you carefully review the descriptions for all contexts (resource group, individual resource, and IVR Profile). For more information about the configuration options, see the other chapters in the Provisioning section of this guide, and see the Genesys Voice Platform 8.0 Configuration Options Reference.

Table 12: Task Summary: Configuring Conferencing


Assign a conference resource to a logical resource group that provides conference service.

See “Configuring Logical Resource Groups” on page 126:1. Create or modify a logical resource group for the Resource

Manager, where the value of the <logical resource group>.service-types option includes conference (see page 133).

2. Set the general conference maximums for the resource group (see the confmaxsize and confmaxcount options).

3. If the resource (for example, Media Control Platform or Call Control Platform) has not already been added to the Resource Manager connections, add it. For more information, see the Genesys Voice Platform 8.0 Deployment Guide chapter about post-installation activities.

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Chapter 4: Configuring Common Features Configuring EMS Reporting

Configuring EMS ReportingTable 13 describes important parameters for EMS Reporting, which you configure in the ems section of the Resource Manager, Media Control Platform, Call Control Platform, Fetching Module, and, if applicable, Cluster Manager Application objects.For general information about EMS Reporting in a GVP deployment, see “Logging and Reporting” on page 60.For information about additional EMS configuration options, see the Genesys Voice Platform 8.0 Configuration Options Reference.

Create an IVR Profile for conference service.

Set the following required parameter:• gvp.service-prerequisite.conference-id (see page 151)Also consider the following IVR Profile options, which determine whether and how conference service will be provided:• gvp.general.application-confmaxsize

• gvp.general.service-type

• gvp.policy.conference-allowed

• gvp.policy.conference-capability-requirements

• gvp.policy.conference-usage-limit and conference-usage-limit-per-session

For more information, see “IVR Profile Configuration Options” on page 141.

If the deployment uses an external conferencing or bridge server, specify the external server IP address on the Media Control Platform.

See the Media Control Platform sip.confserver option.

Verify that conference-related settings on the Media Control Platform and Call Control Platform are suitable.

• For the Media Control Platform, review options in the conference section.

• For the Call Control Platform, verify settings for the Default Conference device profile, and in the mediacontroller and mediactrller configuration sections.

(Optional) Customize the SIP response codes and Resource Manager behavior on error.

On the Resource Manager, customize the value of the rm.conference-sip-error-respcode option.For more information, see Table 64 on page 279.

Table 12: Task Summary: Configuring Conferencing (Continued)




Except where otherwise indicated, all changes to ems configuration options take effect after you restart the component application.

Note: The Resource Manager and Fetching Module do not support upstream metrics, which are application-level logs (relating to VoiceXML or CCXML applications) that are configured to be delivered to the Reporting Server through the Data Collection Sink (DATAC). Configuration parameters in the ems section that relate to DATAC do not apply to the Resource Manager or Fetching Module.

Table 13: Selected EMS Reporting Configuration Options—ems Section

Option Name Description Valid Values and Syntax

(For Media Control Platform and Call Control Platform only)dc.default.logfilter

The filter that determines which logs will be delivered upstream to the Reporting Server for Call Events reporting.Example:On the Media Control Platform, a value of 0,1|176-178|*|4|*|* means that all CRITICAL (0) and ERROR (1) messages for the MCP code modules that have Module IDs in the range 176-178 will be sent to the sink, and all INFO (4) messages for all code modules will be sent as well.

<Levels1>|<ModuleIDs1>|<SpecifierIDs1>[|<Levels2>|<ModuleIDs2>|<SpecifierIDs2>...]

where:• <Levels> is the GVP log

level or a range of log levels.

• <ModuleIDs> identifies the code modules within the tracing component.

• <SpecifierIDs> identifies the specific log messages.

• The pipes (|) are a required part of the syntax, and values within the pipes can be formatted as a range (m-n) or as individual, comma-separated numbers.

For the valid Module and Specifier IDs, see Appendix A, “Module and Specifier IDs” on page 255.The wildcard character (*) means all.Default values: 0-2|*|*

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(For Media Control Platform and Call Control Platform only)dc.default.metricsfilter

The filter that determines which metrics in the Data Collection Sink (DATAC) will be forwarded to the Reporting Server.If the gvp.log.metricsfilter parameter (see page 143) has been set in an IVR Profile, the IVR Profile value will override this default setting, for sessions that execute under this IVR Profile.

<FilterID1>[,<FilterID2>,...]

where:• <FilterID> is a single

Metric ID or a range of Metric IDs. For the valid Metric IDs, see the Genesys Voice Platform 8.0 Metrics Reference.

The wildcard character (*) means all.Default values:• Media Control Platform—0-15,41,52-55,74,136-141

• Call Control Platform—1001,1009,1012-1013,1031,1050,1052,1058-1059

log_sinks A pipe (|)-separated list of the log sinks that will be loaded when the EMS Logging Service initializes.For more information about the log sinks, see “Log Sinks” on page 65.

[MFSINK]|[DATAC]|[TRAPSINK]

Default values: • Resource Manager and

Fetching Module—MFSINK|TRAPSINK

• Media Control Platform and Call Control Platform—DATAC|MFSINK|TRAPSINK

• Cluster Manager—MFSINK

Table 13: Selected EMS Reporting Configuration Options—ems Section (Continued)




logconfig.<Sink Name>

The filter that determines which log messages are sent to the specified sink, where <Sink Name> is one of the following:• MFSINK—the MF Adaptation Sink• DATAC—the Data Collection SinkThe filter is a pipe (|)-separated list of the following sets of parameters:• Log level:

0 = Critical1 = Error2 = Warning3 = Note4 = Info5 = Debug

• Code component that generates the log event.• The specific log message.This configuration option enables GVP to improve EMS Logging performance by prefiltering the log messages. However, Genesys recommends that you retain the default filter setting (all log messages), to ensure that you do not inadvertently omit capturing a type of log message that you later require for tracing purposes.Note: Management Framework can only log to file what is passed to it in accordance with the logconfig.MFSINK filter setting.Changes take effect: Immediately.

<Levels1>|<ModuleIDs1>|<SpecifierIDs1>[|<Levels2>|<ModuleIDs2>|<SpecifierIDs2>...]

where:• <Levels> is the GVP log

level or a range of log levels.

• <ModuleIDs> identifies the code modules within the tracing component.

• <SpecifierIDs> identifies the specific log messages.

• The pipes (|) are a required part of the syntax, and values within the pipes can be formatted as a range (m-n) or as individual, comma-separated numbers.

For the valid Module and Specifier IDs, see Appendix A, “Module and Specifier IDs” on page 255.The wildcard character (*) means all.Default values: See Table 14 on page 110.



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metricsconfig.<Sink Name>

The filter that determines which metrics are sent to the specified sink, where <Sink Name> is one of the following:• MFSINK—the MF Adaptation Sink• DATAC—the Data Collection Sink• TRAPSINK—the SNMP trap sinkThe filter is a comma-separated list of the specific metrics that will be queued and processed.This configuration option enables GVP to improve EMS Reporting performance by prefiltering the metrics. However, Genesys recommends that you do not change the default filter for any sink except MFSINK. For example, you may want to change the settings for the ems.metricsconfig.MFSINK parameter, in order to make a specific set of metrics available to Management Framework, so that these metrics can be sent to Message Server or written to file.Note: Reducing the scope of the metricsconfig.TRAPSINK filter on the Media Control Platform will impact the calculation of VAR summary statistics.Changes take effect: Immediately.




The wildcard character (*) means all.Example:5-8,50-53,70,71 means that metrics with IDs 5, 6, 7, 8, 50, 51, 52, 53, 70, and 71 will be sent to the sink.Default values: See Table 14 on page 110.

(For Media Control Platform and Resource Manager only)ors.reportinginterval

The interval, in seconds, at which accumulated operational reports are submitted to the Reporting Server.

An integer in the range of 1–299.Default value: 60 (seconds)

(For Media Control Platform, Call Control Platform, and Resource Manager)rc.batch_size

The number of upstream messages that the Reporting Client queues before sending them on to the Reporting Server.A higher batch size reduces bandwidth constraints, but at the cost of sending data at larger intervals. See also rc.cdr.batch_size on page 108.

An integer in the range of 1–5000.Default value: 500





(For Media Control Platform, Call Control Platform, and Resource Manager)rc.cdr.batch_size

The number of call detail records (CDRs) to send in a message to the Reporting Server.A higher batch size lessens the demands on network bandwidth and also significantly reduces the transaction overhead on the Reporting Server database, but at the cost of a lag of up to 10 seconds in reporting updates. Batches time out after 10 seconds (in other words, the message is sent after 10 seconds even if the batch is not filled). A batch size of 1 means that CDRs are delivered in near real-time.As a rule of thumb, set the batch size based on the expected call throughput. For example, if a reporting delay of up to 5 seconds is acceptable, set the batch size to the number of CDR updates you expect to receive every 5 seconds, assuming 2 updates per call. Be aware, however, that there could still be a 10-second delay if actual call volumes are less than expected.For high call densities, Genesys recommends retaining the default value of 500.


(For Media Control Platform, Call Control Platform, and Resource Manager)rc.cdr.local_queue_path

The full path to the local SQLite3 database file that serves as the persistent queue for CDRs being submitted to the Reporting Server.Even if GVP component applications share a host, each component must have its own local queue file.

<Path to file>

Default value: $InstallationRoot$\config\cdrQueue.db

(For Media Control Platform, Call Control Platform, and Resource Manager)rc.cdr.msg_broker_uri

The URI of the ActiveMQ message broker that implements Java Message Service (JMS) on the Reporting Server, for CDR reporting.

tcp://<Reporting Server host name or IP address>:<port>

Default value: tcp://$ReportingServerHostname$:61616



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(For Media Control Platform, Call Control Platform, and Resource Manager)rc.local_queue_path

The full path to the local database file that serves as the persistent queue for upstream logging to the GVP Reporting Client.Even if GVP components (for example, Resource Manager or Media Control Platform) share a host, each component has its own local queue file.

<Path to file>

Default value: • For Resource Manager

and Media Control Platform—$InstallationRoot$\config\reportingClientQueue.db

• For Call Control Platform—$InstallationRoot$\config\upstreamQueue_CCP.db

(For Media Control Platform, Call Control Platform, and Resource Manager)rc.msg_broker_uri

The URI of the JMS Message Broker on the Reporting Server.



(For Media Control Platform and Resource Manager only)rc.ors_local_queue_path

The full path to the local SQLite3 database file that serves as the persistent queue for OR statistics being submitted to the Reporting Server.Even if GVP components (for example, Resource Manager or Media Control Platform) share a host, each component has its own local queue file.

<Path to file>

Default value: $InstallationRoot$\config\cdrQueue_rm.db

(For Media Control Platform and Resource Manager only)rc.ors.msg_broker_uri

The URI of the ActiveMQ message broker that implements Java Message Service (JMS) on the Reporting Server, for delivering OR statistics.







Table 14 summarizes the default values for the various logs and metrics filters for the Resource Manager (RM), Media Control Platform (MCP), Call Control Platform (CCP), Fetching Module (FM), and Cluster Manager (CM) Application objects.

Note: Genesys recommends that you do not modify the log filter settings.

trace_flag Flag that indicates whether debug-level logging is enabled. When enabled, debug-level logs will be processed and filtered like logs at other levels.Debug-level logging places a load on the logging infrastructure that degrades performance. Therefore, do not enable it in a production environment unless Genesys Support directs you to do so.Changes take effect: Immediately.

• True

• False

Default value: False



Table 14: Default Log and Metrics Filters

Com-ponent

Option Name (in ems section)*

logconfig.DATAC

logconfig.MFSINK

metricsconfig.DATAC

metricsconfig.MFSINK metricsconfig.TRAPSINK

RM N/A 0-3,5|*|* N/A 0-15,19-41,43,52-56,72-74,76-81,127,129,130,132-141

N/A

MCP 0-2|*|* 0-3,5|*|* * 0-15,19-41,43,52-56,72-74,76-81,127,129,130,132-141

N/A

CCP 0-2|*|* 0-3,5|*|* * 1000-1001,1003-1005,1007-1016,1019-1021,1024,1027-1036,1039-1045,1048-1050,1052-1054,1056,1058-1062

*

* For descriptions of the configuration options, see Table 13 on page 104.

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Chapter 4: Configuring Common Features Configuring Logging

Configuring LoggingTable 15 describes the parameters for logging that are most commonly customized. Table 16 on page 115 summarizes the default values for these options in GVP. The options are in the log configuration section of each GVP component Application.Configure the options for each component in the Genesys Administrator on the Provisioning > Environment > Applications > <GVP Application> >

Settings tab. For the detailed steps to configure option settings, see Viewing or modifying GVP configuration parameters, page 78.Changes take effect immediately.The Application Templates do not expose all the logging parameters that are standard in Genesys applications. The Settings tab and its metadata (which are also available in the Genesys Voice Platform 8.0 Configuration Options Reference) therefore do not describe all the parameters that determine the logging behavior of GVP applications. For more information about the additional, standard logging options, see the Log Section in the chapter about common configuration options in the Framework 8.0 Configuration Options Reference Manual.

FM N/A 0-3,5|*|* N/A 0-15,19-41,43,52-56,72-74,76-81,127,129,130,132-141

N/A

CM N/A *|*|* N/A * N/A

Table 14: Default Log and Metrics Filters (Continued)

Com-ponent

Option Name (in ems section)*

logconfig.DATAC

logconfig.MFSINK

metricsconfig.DATAC

metricsconfig.MFSINK metricsconfig.TRAPSINK

* For descriptions of the configuration options, see Table 13 on page 104.



Table 15: Selected Configuration Options—log Section

Parameter Name Description Valid Values and Syntax

all A comma-separated list of the output destinations to which the Application (GVP process) sends all log events.Setting log.verbose to all and this parameter (log.all) to network enables an application to send log events of the Standard, Interaction, and Trace levels to Message Server. With this setting, Debug-level log events are not sent to Message Server and are not stored in the Log Database.

• stdout—Log events are sent to the Standard output (stdout).

• stderr—Log events are sent to the Standard error output (stderr).

• network—Log events are sent to Message Server, which can reside anywhere on the network. Message Server stores the log events in the Log Database.

• memory—Log events are sent to the memory output on the local disk. This is the safest output in terms of application performance.

• <filename>—Log events are stored in a file with the specified name. The default path for the file is the working directory of the application.

standard A comma-separated list of the output destinations to which the application (GVP process) sends log events of the Standard level.

interaction A comma-separated list of the output destinations to which the application (GVP process) sends log events of the Interaction level and higher (that is, Standard and Interaction levels).

trace A comma-separated list of the output destinations to which the application (GVP process) sends log events of the Trace level and higher (that is, Standard, Interaction, and Trace levels).

debug A comma-separated list of the output destinations to which the application (GVP process) sends log events of the Debug level and higher (that is, Standard, Interaction, Trace, and Debug levels).

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expire (Applicable only if log output is configured to be sent to a log file.)Specifies the criteria for determining when log files (segments) expire and are deleted.

• false—No expiration. All generated segments are stored.

• <number> file|<number>—A number in the range of 1–100 that specifies the maximum number of log files to store.

• <number> day—A number in the range of 1–100 that specifies the maximum number of days before log files are deleted. (Not applicable for Reporting Server.)

message_format The log record header format that an Application uses when writing logs in the log file. Using compressed log record headers improves application performance and reduces the size of the log file. When message_format=short (compressed headers):• A header of the log file or the log file

segment contains information about the application (such as the application name, application type, host type, and time zone), whereas individual log records within the file or segment do not contain this information.

• Log message priority is abbreviated to Std, Int, Trc, or Dbg (instead of Standard, Interaction, Trace, or Debug, respectively).

• The message ID does not contain the prefix GCTI or the application type ID.

• short—The Application uses compressed headers when writing log records in the log file.

• full—The Application uses complete headers when writing log records in the log file.

Log record examples:• Full format: 2002-05-07T18:11:38.196 Standard localhost cfg_dbserver GCTI-00-05060 Application started

• Short format:2002-05-07T18:15:33.952 Std 05060 Application started

Table 15: Selected Configuration Options—log Section (Continued)




segment (Applicable only if log output is configured to be sent to a log file.)Specifies the mode of measurement and maximum size for a log file segment. If the current log segment exceeds the size set by this option, the file is closed and a new log file is created.

• false—No segmentation is allowed.

• <number> KB|<number>—The maximum segment size, in kilobytes. The minimum segment size is 100 KB.

• <number> MB—The maximum segment size, in megabytes.

• <number> hr—The number of hours for the segment to stay open. The minimum time period is 1 hour. (Not applicable for Reporting Server.)

time-format The format in which the log file presents the time when the application generated the log record.

• time—The time string is formatted according to the HH:MM:SS.sss format (hours, minutes, seconds, milliseconds).

• locale—The time string is formatted according to the locale of the system.

• ISO8601—The date in the time string is formatted according to the ISO 8601 format. Fractional seconds are given in milliseconds.Example:2001-07-24T04:58:10.123



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Table 16 provides the default values for options in the log configuration section that are commonly modified.

verbose Specifies the minimum level of log events that will be generated.In descending order of priority, the log event levels are:• Standard• Interaction• Trace• Debug

• all—All log events.• debug—Log events of all

levels (same as all).• trace—Log events of the

Standard, Interaction, and Trace levels.

• interaction—Log events of the Standard and Interaction levels.

• standard—Log events of the Standard level only.

• none—No output will be generated.



Table 16: Default Values for Selected log Options

Option Name

Default Value

Resource Manager


Call Control Platform

Fetching Module

Reporting Server

Cluster Manager

all Empty ..\logs\MCP ..\logs\ccp $InstallationRoot$\logs\fm,stdout

Empty Empty

debug ..\logs\ResourceMgr

..\logs\MCP ..\logs\ccp Empty Empty ..\logs\ClusterMgr

expire 20 (files) 10 (files) 20 (files) 10 (files) false 20 (files)

interaction ..\logs\ResourceMgr,network

..\logs\MCP ..\logs\ccp,network

..\logs\mcp_metricsfile,network

Empty ..\logs\ClusterMgr,network

message_format

short short [Not exposed in template]

short short short

segment 10000 KB 10000 KB 10000 KB 10000 KB false 10000 KB

standard ..\logs\ResourceMgr,network

..\logs\MCP ..\logs\ccp,network

..\logs\MCP,network

network ..\logs\ClusterMgr,network


Chapter 4: Configuring Common Features Customizing SIP Responses

Customizing SIP ResponsesThis section lists the configuration options that enable you to customize the SIP responses and alarms that the Resource Manager, Media Control Platform, and Call Control Platform signal for certain events and conditions.For more information about the SIP response codes that are generated, and how the following configurable options relate to them, see Table 64 on page 279. For more information about all the configuration options, see the Genesys Voice Platform 8.0 Configuration Options Reference.To customize GVP behavior in response to error conditions and other events, consider the following options.• Resource Manager:

rm.conference-sip-error-respcode

rm.options_response_contenttype

rm.options_response_msg_body

rm.resource-no-match-respcode

rm.resource-unavailable-respcode

rm.suspend-mode-respcode

<gateway resource group>.noresource-response-code

• IVR Profile:gvp.policy.ccxml-usage-limit-exceeded-respcode

gvp.policy.ccxml-usage-limit-exceeded-set-alarm

gvp.policy.conference-forbidden-respcode

gvp.policy.conference-forbidden-set-alarm

gvp.policy.conference-usage-limit-exceeded-respcode

gvp.policy.conference-usage-limit-exceeded-set-alarm

gvp.policy.dialing-rule-forbidden-respcode

gvp.policy.dialing-rule-forbidden-set-alarm

time-format

time time time time time time

trace ..\logs\ResourceMgr

..\logs\MCP ..\logs\ccp Empty Empty ..\logs\ClusterMgr

verbose standard interaction standard interaction trace standard

Table 16: Default Values for Selected log Options (Continued)

Option Name

Default Value

Resource Manager


Call Control Platform

Fetching Module

Reporting Server

Cluster Manager

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Chapter 4: Configuring Common Features Configuring Session Timers and Timeouts

gvp.policy.external-sip-forbidden-respcode

gvp.policy.external-sip-forbidden-set-alarm

gvp.policy.external-sip-usage-limit-exceeded-set-alarm

gvp.policy.inbound-usage-limit-exceeded-set-alarm

gvp.policy.outbound-call-forbidden-respcode

gvp.policy.outbound-call-forbidden-set-alarm

gvp.policy.outbound-usage-limit-exceeded-set-alarm

gvp.policy.transfer-forbidden-respcode

gvp.policy.transfer-forbidden-set-alarm

gvp.policy.usage-limit-exceeded-respcode

gvp.policy.usage-limit-exceeded-set-alarm

gvp.policy.voicexml-dialog-forbidden-respcode

gvp.policy.voicexml-dialog-forbidden-set-alarm

gvp.policy.voicexml-usage-limit-exceeded-respcode

gvp.policy.voicexml-usage-limit-exceeded-set-alarm

• Media Control Platform:sip.sendalert

sip.copyunknownheaders

• Call Control Platform:ccpccxml.defaultrejectcode

ccpccxml.sip.send_progressing

session.copy_unknown_headers

sip.copyunknownheaders

sip.OPTIONS.header.Accept

sip.OPTIONS.header.Accept-Encoding

sip.OPTIONS.header.Accept-Language

sip.OPTIONS.header.Allow

To further customize the SIP response code for specific situations, use the <hints> attribute of the <redirect> and <reject> tags—the responseCode property of the hints object specifies the response code to be used.

Configuring Session Timers and TimeoutsThis section describes two kinds of timeouts that determine the length of a session and affect responses to service requests:• The session inactivity timers, expiry timers, and timeouts that the Resource

Manager uses to manage sessions (see “Resource Manager Session Timers”).

• Additional timeouts that are set at the resource level or specified in SIP or HTTP requests (see “Additional Timeouts” on page 120).



Resource Manager Session TimersTable 17 summarizes the configuration options that determine the session timers that the Resource Manager uses to manage sessions, in the order in which the Resource Manager applies them. Configure these options on the IVR Profile, Tenant, or Application (Media Control Platform, Call Control Platform, Resource Manager) objects, as applicable for your deployment.The Resource Manager adds a Session-Expires header to initial INVITE requests if one is not present, and if the request does not contain the timer option in the Supported header. The value of the Session-Expires header is the configured value of the applicable session inactivity timer, except under the conditions described in “Session-Expires Header” on page 29.For more information about how the Resource Manager uses expiry timeouts, see “Session Management” on page 27.

Table 17: Session Timer Configuration Options

Section.Parameter Name


IVR Profile

gvp.general.sip.sessiontimer

The timeout value, in seconds, for the SIP session that executes for this IVR Profile. If the Resource Manager receives no SIP messages associated with this call leg within the timeout interval, the Resource Manager considers the call leg to have ended.For the call leg associated with this IVR Profile, the value that you configure for this sip.sessiontimer parameter overrides session expiry timeouts that are set at the level of the tenant, the resource, and the Resource Manager.

Any positive integer.Default value: Empty

Environment Tenant

gvp.general.sip.sessiontimer

The timeout value, in seconds, for the SIP session that executes for the IVR Profile. If the Resource Manager receives no SIP messages associated with this call leg within the timeout interval, the Resource Manager considers the call leg to have ended.For the call leg associated with the IVR Profile for this tenant, the value that you configure for this sip.sessiontimer parameter overrides session expiry timeouts that are set at the level of the resource and the Resource Manager.


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Media Control Platform/Call Control Platform

sip.min_se

The minimum value of the Session-Expires header, in seconds, that the SIP stack will accept from a UAC (User Agent Client).


sip.sessionexpires

The default timeout interval, in seconds, for Media Control Platform or Call Control Platform sessions. If no re-INVITES are sent or received within the timeout period, the session expires.If a different timeout has been set for a particular VoiceXML or CCXML application, it overrides the value of this sip.sessionexpires parameter.


Resource Manager

proxy.sip.min_se

The minimum value of the Session-Expires header, in seconds, that the Resource Manager will accept.If an incoming SIP request contains a Session-Expires header with a value that is less than sip.min_se, the Resource Manager rejects the INVITE request with a 422 (Session interval too small) response.Changes take effect: After restart.

Any unsigned integer.Default value: 90

proxy.sip.sessionexpires

The timeout value, in seconds, for each SIP session (call leg) that the Resource Manager handles.If a different timeout has been set for a particular resource or XML application, it overrides the Resource Manager session expiry timeout for the applicable session.Changes take effect: After restart.


Table 17: Session Timer Configuration Options (Continued)





Additional TimeoutsThe following timeouts and timers are also important for GVP behavior.• Resource Manager, Media Control Platform, and Call Control Platform:

sip.timer.ci_proceeding—The timeout for a client transaction that is in progress. If a final response is not received within the timeout period, the client transaction is considered terminated. (Default is 120000 ms, or 120 seconds.)

• Media Control Platform:mpc.rtp.timeout—The timeout for the RTP stream. (Default is 60000 ms.)sessmgr.acceptcalltimeout—The timeout for the platform to accept inbound calls, after alerting is issued. (Default is 20000 ms.)sessmgr.maxincalltime—The maximum call time for inbound calls. (Default is 0—disabled.)sip.hfdisctimer—The timeout to terminate a SIP hookflash transfer. (Default is 5000 ms.)stack.connection.timeout—The connection timeout for the MRCP Client stack to establish a TCP connection to the MRCP server. (Default is 10000 ms.)stack.client.timeout—The connection timeout for the MRCP Client to receive a response from the MRCP server. (Default is 10000 ms.)vxmli.default.connecttimeout—The default value of the connecttimeout attribute for bridge or consultation transfers, if not provided. (Default is 30000 ms.)vxmli.initial_request_fetchtimeout—The fetch timeout for the initial VoiceXML page. (Default is 30000 ms.)

registrar.sip.registrar.maxexpirytime

The maximum expiry time, in seconds, of this registrar. If the client requests an expiry time greater than this value, this sip.registrar.maxexpirytime value is the value that will be returned.


registrar.sip.registrar.minexpirytime

The minimum expiry time, in seconds, of this registrar. If the client requests an expiry time smaller than this value, the request will be rejected, with this value in the Min-Expires header.


Table 17: Session Timer Configuration Options (Continued)



User’s Guide 121


vxmli.max_script_time—The maximum time allowed for each script or ECMAScript expression to execute. (Default is 2000 ms.)

• Call Control Platform:ccxmli.fetch.timeout—The default timeout for the fetch of the initial page to complete. (Default is 30 seconds.)



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Chapter

5 Configuring the Resource ManagerThis chapter describes the Resource Manager configuration requirements in your Genesys Voice Platform (GVP) deployment.It contains the following sections:

Task Summary: Configuring the Resource Manager, page 123Important Resource Manager Configuration Options, page 124Configuring Logical Resource Groups, page 126Enabling High Availability, page 134

Task Summary: Configuring the Resource Manager

Table 18 summarizes the configuration steps and options to implement Resource Manager functionality in your GVP deployment.


Chapter 5: Configuring the Resource Manager Important Resource Manager Configuration Options

Important Resource Manager Configuration Options

This section describes the key configuration options that you either must or may want to customize.Configure the options in the Genesys Administrator on the Provisioning >

Environment > Applications > <Resource Manager> > Settings tab. For the detailed steps to configure option settings, see Viewing or modifying GVP configuration parameters, page 78.Except where otherwise indicated, all changes to Resource Manager parameters take effect after you restart the Resource Manager.

Table 18: Configuring the Resource Manager


Set up the Resource Manager to function as SIP Proxy, SIP Registrar, and resource monitor and manager.

See “Configuring SIP Communications and Routing” on page 86.To secure SIP communications between the Resource Manager and the other GVP components, ensure that you specify a transport for the Transport Layer Security (TLS) protocol and a secure routeset for outbound calls.

(Optional) Enable High Availability. • Configuring Resource Manager High Availability, page 134.• Setting up the cluster mode execution environment, page 136.

Provision the GVP resources. See “Configuring Logical Resource Groups” on page 126.

Provision the IVR Profiles. See Chapter 6 on page 139.

Configure conferencing. See “Enabling Conference Services” on page 102.

Configure EMS Reporting. See “Configuring EMS Reporting” on page 103.

Customize logging. See “Configuring Logging” on page 111.

Customize session management behavior and performance.

See “Configuring Session Timers and Timeouts” on page 117.See also parameters in the proxy section that specify parameters such as numbers of threads and connections.

Customize Resource Manager messaging.

See “Customizing SIP Responses” on page 116 and Table 64 on page 279.

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Chapter 5: Configuring the Resource Manager Important Resource Manager Configuration Options

The Resource Manager configuration options are in the following configuration sections:• ems (see Table 13 on page 104)—Parameters determine EMS Reporting

behavior.• log (see “Configuring Logging” on page 111)—Parameters determine

behavior for Management Framework logging.• rm—Parameters determine the behavior of the Resource Manager in its role

as manager of GVP services.• proxy—Parameters determine the behavior of the Resource Manager in its

role as SIP Proxy and session manager.• registrar—Parameters determine the behavior of the Resource Manager

in its role as SIP Registrar.• monitor—Parameters support the Resource Manager in its role as manager

of GVP resources.• cmserviceagent—The only parameter (cmport) supports High Availability,

by configuring the port for communicating with the Cluster Manager.Table 19 provides information about important Resource Manager parameters that are not described in Chapter 4 on page 85. Table 19 provides parameter descriptions as well as the default parameter values that are preconfigured in the Resource Manager Application object. For information about all the available configuration options for the Resource Manager, see the Genesys Voice Platform 8.0 Configuration Options Reference.

Table 19: Selected Resource Manager Configuration Options


rm Section

cluster_ip (Required for warm active-standby mode, for High Availability) The virtual IP address that the external load balancer will use to provide transport-level redundancy.If no value is set, the Resource Manager is not running in any form of active-standby mode.For more information, see Configuring Resource Manager High Availability, page 134.

<IP address>



Chapter 5: Configuring the Resource Manager Configuring Logical Resource Groups

Configuring Logical Resource GroupsFor each type of service that GVP provides (VoiceXML, CCXML, Conference, or Gateway), you must create and configure a logical resource group that Resource Manager will use as its resource pool. You must create a resource group for each type of service, even if there is only one resource available to provide that service (in other words, the group has a single member). Use the Genesys Administrator Provisioning > Voice Platform > Resource

Management > <Resource Manager> > Manage RM Resources wizard to create, modify, or view settings for the resource group and to specify the resources

sip-header-for-dnis The header from which the Resource Manager will retrieve the DNIS, to identify which IVR Profile to use.Ensure that the value you specify is consistent with Media Gateway behavior, so that the INVITE messages that SIP Server forwards to the Resource Manager have the DNIS information in the expected header.• If the value of this parameter is History-Info

but there is no History-Info header in the SIP INVITE, the Resource Manager picks up the DNIS from the To header.

• If the value of the specified header in the SIP INVITE is not a valid DNIS, the Resource Manager cannot map the SIP request to an IVR Profile and defaults to the next behavior to select the IVR Profile (see “Mapping the Call to an IVR Profile” on page 30).

Changes take effect: Immediately.

• To

• Request-Uri

• History-Info

Default value: History-Info

monitor Section

sip.proxy.optionsinterval

The interval, in milliseconds, at which the Resource Manager sends OPTIONS messages to a healthy resource to determine if the resource is alive.


sip.proxy.unavailoptionsinterval

The interval, in milliseconds, at which the Resource Manager sends OPTIONS messages to a dead resource to determine if the resource is alive.


Table 19: Selected Resource Manager Configuration Options (Continued)


User’s Guide 127


that belong to each group. For details about using the Manage RM Resources wizard, see Configuring logical resource groups.You identify the actual resource hosts and applications in the connections that you configure for the Resource Manager. For more information, see the Genesys Voice Platform 8.0 Deployment Guide chapter about post-installation activities.The following procedure provides the detailed steps to use the Manage RM Resources wizard.

Procedure:Configuring logical resource groups

Purpose: To create or modify the property information that is shared by a logical group of GVP resources managed by a particular Resource Manager.

Except where indicated otherwise, all changes to logical resource group configurations take effect immediately.

Prerequisites

• The GVP Application objects have been installed, as described in the Genesys Voice Platform 8.0 Deployment Guide.




Start of procedure

1. In the Genesys Administrator, go to the Provisioning > Voice Platform >

Resource Management panel (see Figure 7).

Figure 7: The Resource Management Panel

2. Select the Resource Manager for which you want to configure the logical resource group.

3. In the Tasks panel, click Manage RM Resources.The Step 1: Group List page of the GVP Manage <Resource Manager> Resources wizard displays.

4. On the Step 1: Group List page, do one of the following:• To create a new group, click New.

The New Group dialog box displays.• To modify the configuration parameters for an existing group, select

the group name and click Modify.The Modify Group dialog box displays.

• To modify the resources that are assigned to an existing group, click Next. Continue at Step 6.

Tip: Do not click the name of the Resource Manager, or you will link to the Provisioning > Environment > Applications > <Resource Manager> > Configuration tab. Rather, click anywhere else in the applicable row.

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• To delete an existing group, click Delete. After you confirm the deletion, click through the remainder of the Wizard to Finish.

Note: Deleting a group does not delete the resources themselves from the GVP deployment. However, connections from the Resource Manager to members of the deleted resource group are removed. In GVP 8.0, a resource cannot be a member of more than one group. Deleting a group therefore means that connections to all the resources in that group are removed.

5. Use the text boxes and drop-down lists in the New Group or Modify Group dialog box to specify the configuration parameters for the group. You cannot change the name of an existing group. For more information about the logical group configuration parameters, including the types of services, see Table 20 on page 131.After you have specified all the required options, click OK. You are returned to the Step 1: Group List page.

6. To add resources to the group, access the Step 2: Group Resources page in one of the following ways:• With the required group selected on the Step 1: Group List page, click

Next.

• With the required group selected on the Step 1: Group List page, click Step 2: Group Resources in the Steps panel.

7. On the Step 2: Group Resources page, select or clear the check boxes in the Assigned column to add resources to or remove resources from the group. Except for gateway services, the list of available resources (the Application list) includes only those resources in your deployment that support the service type(s) you specified for the group in Step 5. The list of gateway resources will include all T-Servers (including SIP Servers) in the Genesys deployment. Ensure that you select only the applicable SIP Servers to add to the gateway resource group. You can also add Resource Access Point Applications to a gateway resource group.When you select a check box to add the resource, the Address-of-record and Max. Ports fields become active.

8. Specify the required resource-specific parameters:a. The Address-of-record (aor parameter) for the resource.

The aor parameter is a comma-separated list of the public addresses for the physical resource, in the following format:<protocol>:<ip1>:<port1>[,<protocol>:<ip2>:<port2>,...]

where:• <protocol> is sip or sips.• <ip> is the address of record (AOR) of the resource host.



• <port> is the SIP port on the resource host.For example: sip:ResourceHost.company.com:5060

b. The Max. Ports (port-capacity parameter) for the resource.The port-capacity parameter specifies the maximum number of concurrent requests the resource is capable of handling. The value can be any unsigned integer.

9. Click Next, or else click Step 3: Confirmation in the Steps panel, to access the Step 3: Confirmation page.

10. To confirm and save the changes, click Finish.

End of procedure

Next Steps

• If required, configure the noresource-response-code option in the <gateway resource group> section of the Resource Manager Application object (see page 132).The default behavior for the Resource Manager with regard to gateway resources is not to retry failed requests. To configure the Resource Manager to automatically retry other resources in a gateway group, specify the required SIP failure response codes in the noresource-response-code option. This option does not appear in the Resource Management wizard. Configure the option on the Provisioning > Environment > Applications > <Resource Manager> > Options tab.

Table 20 provides information about the Resource Manager parameters for logical groups. Configure these parameters using the Resource Management wizard (see Step 5 on page 129) or else on the Provisioning > Environment > Applications > <Resource Manager> > Options tab, in the configuration section for the applicable logical resource group.

User’s Guide 131


Table 20: Logical Group Section Configuration Options


capability A list of name-value pairs that identify the capabilities that are supported by resources in the logical resource group.Items in the name-value pair list are separated by a semi-colon. The value side of each name-value pair can itself be a comma-separated list of capabilities. Each set of values must be unique.The Resource Manager will direct interactions to the resource group only if the values in this option (or a subset of the name-value pairs) exactly match the capability requirements that are specified in the applicable VoiceXML or CCXML application.For example, say the value of this option is set to lang=en-US;grammar=grxml,gsl for a VoiceXML group.• If the voicexml-capability-requirement

attribute in the gvp.policy annex of the VoiceXML application is specified as "lang=en-US", then the Resource Manager will send the call to the group.

• If the capability is specified in the VoiceXML application as "lang=en-US,en-IN", then the Resource Manager will either send the call to another VoiceXML group that does support the capability, or it will reject the call (rm.resource-no-match-respcode—see page 283).

<cap_NameA>=<cap_ValueA>[;<cap_NameB>=<cap_ValueB>;...]

where:• <cap_NameX> is the name

of the capability.• <cap_ValueX> is a

comma-separated list of values.

Example:lang=en-US;grammar=grxml,gsl (for a VoiceXML group)Default value: Empty

load-balance-scheme The distribution algorithm that the Resource Manager will use to select a resource within this logical resource group.

• round-robin

• least-used

Default value: round-robin



monitor-method The method the Resource Manager will use to determine if the physical resources belonging to the logical resource group are alive and healthy.

• option—Resource Manager will use SIP OPTIONS messages.

• none—Resource Manager will not monitor resource health. The Resource Manager assumes that resources in this group are always alive.

Default value: None

noresource-response-code

The gateway resource failure response codes that will cause the Resource Manager to retry the request on other gateway resources in the group. The Resource Manager receives the responses from SIP Server.The parameter value is a list of response codes, separated by semi-colons.• If the Resource Manager receives a failure

response that matches one of the codes in the list, it keeps retrying the request on other gateway resources in the logical resource group (in other words, normal Resource Manager behavior in relation to resource groups).

• If the Resource Manager receives a failure response that does not match any of the codes in the list, it forwards the response to the UAC and does not retry the request.

This parameter enables more fine-grained management of gateway resources, provided the gateway has been configured to differentiate its responses.

Any valid SIP response code.Default value: EmptyExample:Say <gateway resource group>.noresource-response-code=503;504.The gateway has been configured to provide the following response codes:• 503 for requests that fail

because capacity has been reached on the gateway resource

• 486 for calls that fail because the callee is busy.

A request fails because:• The gateway has reached

its capacity (503 response)—The Resource Manager retries the request on another gateway in the group.

• The callee is busy (486 response)—The Resource Manager simply forwards the response to the requesting Media Control Platform or Call Control Platform, without retrying the request.

Table 20: Logical Group Section Configuration Options (Continued)


User’s Guide 133


service-types The types of service that are provided by resources in this logical resource group.To specify multiple types of service (for example, CCXML and Conference), hold down the Ctrl key while selecting additional service types. Resources can be assigned to the group only if they support all the service types that you specify in this parameter.Changes take effect: After restart

• voicexml—Voice application services provided by Media Control Platform resources.

• ccxml—Call control application services provided by Call Control Platform resources.

• gateway—Network gateway services provided by Resource Access Point resources.

• conference—Conference services, which can be provided by Media Control Platform and Call Control Platform resources.

• external-sip—SIP services provided by an external SIP proxy.

port-usage-type Determines which SIP dialogs the Resource Manager will consider when calculating the current usage on each resource, for resource management purposes.Current usage is defined as the outstanding number of established SIP dialogs on a resource plus the current pending requests on the resource. The SIP dialogs that are included in the calculation are:• For in-and-out—SIP dialogs originated

from and directed to the resource.• outbound—SIP dialogs directed to the

resource.Changes take effect: After restart

• in-and-out—For gateway resources only (service-type=gateway).

• outbound—For all non-gateway resources.

Default value:

resource-confmaxsize (Applicable only for groups where service-type=conference.)The maximum conference size (number of participants) supported by resources in this logical resource group.

Any unsigned integer.Default value:




Chapter 5: Configuring the Resource Manager Enabling High Availability

Enabling High AvailabilityGVP uses Windows Network Load Balancing (NLB) clustering to provide High Availability (HA) for the Resource Manager.A Cluster Manager on each Resource Manager host monitors the Network Interface Cards (NICs) in cluster mode to determine when network errors occur.The following procedure provides an overview of the steps to create an NLB cluster and to configure the Resource Manager Application to operate as a node in a cluster.For more detailed instructions about creating and configuring the cluster, see the appendix about NLB clustering in the Genesys Voice Platform 8.0 Deployment Guide.

Procedure:Configuring Resource Manager High Availability

Purpose: To provide a high-level description of the steps to enable High Availability for the Resource Manager in a GVP deployment.

Prerequisites

• Two Resource Manager Applications have been configured to function in stand-alone mode on two NLB cluster node machines, which reside in the same subnet. Each NLB cluster node machine has at least two NICs.

• An IP address has been allocated to be used as the virtual IP address.

resource-confmaxcount

(Applicable only for groups where service-type=conference.)The maximum number of concurrent conferences supported by each resource in this logical resource group.

Any unsigned integer.Default value:



User’s Guide 135


Start of procedure

1. Set up the Windows NLB service on each Resource Manager host.See the Genesys Voice Platform 8.0 Deployment Guide for details about the required values you must set for the cluster parameters (including the virtual IP address for the cluster), host parameters, port rules, and other properties.

2. In the Genesys Administrator, create and configure a GVP Cluster Manager Application object for each Resource Manager Application in the cluster. At a minimum, you must specify the following GVP Cluster Manager parameters:• A space-separated list of IDs of cluster members (Cluster.members

option). The IDs correspond to the NLB unique host identifier (priority) number that you specified for each NLB cluster machine in the NLB service properties (Step 1). The default is 1 2.

• For each cluster member, the IP address and TCP port on which the cluster member can be reached (Cluster.member.<x> option).• The IP address is the private IP address of the host, not the virtual

IP address for the cluster. • For each member.<x>, the port number must correspond to the port

specified in the Cmcomm.tcpbondinglocalport option of the Cluster Manager for that member. The default port is 9801.

• The member ID (Cluster.mymemberid option), which is the ID of the Cluster Manager instance, and corresponds to an ID in the list of cluster members.

• The path to the NLB.bat executable (Cluster.NLBScriptPath option). The default path is <Cluster Manager installation directory>\bin\NLB.bat.

• If necessary, also specify the value for the TCP port where this cluster manager is listening for the Resource Manager to connect (Cmservicemgrbase.serverport option). This port must correspond to the port specified in the Resource Manager cmserviceagent.cmport option. The default port is 6001.

3. (Optional) Specify the NICs that need to be monitored. Genesys recommends that you specify the list of NICs to be monitored if the machines also have other NICs that are not involved in the NLB cluster.a. On the Options tab of the Cluster Manager Application, create the gvp

section.b. Specify the required nic.eth<x> options, starting with nic.eth0, as

name-value pairs, where the value is the MAC address of the NIC card.4. On each Cluster Manager Application, create a connection to Message

Server, to enable logging.



5. Configure each Resource Manager Application to execute in cluster mode. Specify the following options:

6. In a text editor, edit the <Resource Manager installation directory>\bin\init.bat file on each cluster machine:• Replace the IP addresses with the virtual IP address.• If necessary, replace the <member id> part of <IP address>:

<member id> with the correct member ID, as defined in Step 2.7. In a text editor, edit the NLB.bat file on each cluster machine:

• Replace the IP address with the virtual IP address.

End of procedure

Next Steps

• Setting up the cluster mode execution environment

Procedure:Setting up the cluster mode execution environment

Purpose: To start the Resource Manager in HA mode.

Prerequisites

• The NLB cluster and the Resource Manager Applications have been configured for HA (see Configuring Resource Manager High Availability, page 134).

Section.Option Valuerm.cluster_ip <Virtual IP address for the cluster>

proxy.sip.transport.0 transport0 udp:any:5060

proxy.sip.transport.1 transport1 tcp:<Virtual IP address for the cluster>:5060

proxy.sip.transport.2 transport2 tls:<Virtual IP address for the cluster>:5061 cert=$InstallationRoot$\config\x509_certificate.pem key=$InstallationRoot$\config\x509_private_key.pem

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Start of procedure

1. On each cluster machine, execute the modified <Resource Manager installation directory>\bin\init.bat. This disables other load-balancing and ensures that Cluster Manager, which is monitoring Resource Manager health, is the entity enabling load-balancing.

2. Start the Cluster Manager–Resource Manager pairs on each NLB cluster node. (In other words, two instances of Cluster Manager and two instances of Resource Manager will be operating.)

3. Verify that each Cluster Manager–Resource Manager instance is running as a separate Genesys Application.

End of procedure



User’s Guide 139

Chapter

6 Provisioning IVR ProfilesIVR Profiles are the Voice Extensible Markup Language (VoiceXML) and Call Control Extensible Markup Language (CCXML) applications that control interactions with external customers. This chapter describes how to provision IVR Profiles for GVP.It contains the following sections:

Provisioning IVR Profiles for GVP, page 139IVR Profile Configuration Options, page 141Mapping IVR Profiles to Dialed Numbers, page 151Specifying Tenant Environment Settings, page 153

Provisioning IVR Profiles for GVPThe summary procedure in this section provides an overview of the steps to provision IVR Profiles for GVP.

Procedure:Provisioning IVR Profiles

Purpose: To set up GVP so that it will use specified VoiceXML or CCXML applications to control interactions that use DNs for which GVP provides service.

Prerequisites

• The VoiceXML or CCXML applications have been created and are available on a network path accessible to GVP. For information about developing the applications, see the Genesys Voice Platform 8 VoiceXML 2.1 Help and the Genesys Voice Platform 8 CCXML


Chapter 6: Provisioning IVR Profiles Provisioning IVR Profiles for GVP

Reference Manual. For information about using Genesys Composer to develop the applications, see Composer Voice 8 Help.


Start of procedure

1. On the Provisioning > Voice Platform > IVR Profile tab of the Genesys Administrator, create an IVR Profile object.The IVR Profile name that you specify is used by the Resource Manager to identify the context of the session. For more information, see “Application Identifiers” on page 72.For more information about creating an IVR Profile, see the chapter about post-installation activities in the Genesys Voice Platform 8 Deployment Guide.

2. Configure the IVR Profile.a. Use the Provisioning > Voice Platform > IVR Profile > <IVR

Profile Property> > Settings tab and Options tab to view and modify values for the IVR Profile configuration options.For detailed information about the IVR Profile configuration options, see “IVR Profile Configuration Options” on page 141.

b. (Optional) To specify database retention policies for the IVR Profile, to override the Reporting Server defaults for the overall GVP deployment, create a new configuration section named dbmp on the Provisioning > Voice Platform > IVR Profile > <IVR Profile> >

Options > Annex list tab, and configure additional rs.db.retention.<type of data> options as required. For more information, see the description for the equivalent rs.db.retention.<type of data>.default options on page 209.

c. (Optional) Use the Provisioning > Voice Platform > IVR Profile > <IVR Profile Property> > Number Mapping tab to map the IVR Profile to a DNIS range. You can also specify this mapping in other ways (see “Mapping IVR Profiles to Dialed Numbers” on page 151).

3. Repeat Steps 1 and 2 as required for all the voice and call control applications that you want to provision.

4. (Optional) On the Provisioning > Voice Platform > Dialed Number Mapping tab, map the IVR Profiles to DNIS ranges. For more information, see “Mapping IVR Profiles to Dialed Numbers” on page 151.

5. Specify the default IVR Profile for GVP:a. Go to the Provisioning > Environment > Tenants > Environment

[tenant] > Options > Annex list tab.b. If necessary, create the gvp.general configuration section.

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Chapter 6: Provisioning IVR Profiles IVR Profile Configuration Options

c. Specify the name of the default IVR Profile as the value for the gvp.general.default-application parameter.

6. On the Provisioning > Environment > Tenants > Environment [tenant] > Options > Annex list tab, specify other tenant settings for using IVR Profiles. For more information, see “Specifying Tenant Environment Settings” on page 153.

Note: GVP 8.0 supports single tenancy only. Therefore, the Environment tenant is the only tenant.

End of procedure

IVR Profile Configuration OptionsThe IVR Profile configuration options determine the type of service the IVR Profile will provide, as well as its operating parameters.Configure the IVR Profile parameters in the Genesys Administrator:• Configure the parameters in the gvp.service-parameters section and, if

applicable, the dbmp section on the Provisioning > Voice Platform > IVR

Profile > <IVR Profile> > Options > Annex list tab. • Configure all the other IVR Profile parameters on the Provisioning >

Voice Platform > IVR Profile > <IVR Profile> > Settings tab.For more information about using the Genesys Administrator to add or modify configuration sections and options, see Viewing or modifying GVP configuration parameters, page 78.The following tables describe the IVR Profile configuration options, by configuration section:• gvp.general Section—Table 21 on page 142• gvp.log Section—Table 22 on page 143• gvp.policy Section—Table 23 on page 143• gvp.policy.dialing-rules—Table 24 on page 149• gvp.service-parameters Section—Table 25 on page 150• gvp.service-prerequisite Section—Table 26 on page 151

Note: All changes to IVR Profile configuration options take effect with the next session that uses the IVR Profile.

gvp.general SectionTable 21 describes the parameters in the gvp.general section.



gvp.log SectionTable 22 describes the parameter in the gvp.log section. This parameter enables you to specify the metrics filters that all resources will use for logging data, when they provide services for the IVR Profile.

Table 21: IVR Profile Configuration Options—gvp.general Section


application-confmaxsize

(For Conference only) The maximum number of participants in the conference.This setting does not override conference size maximums that are configured for the Resource Manager logical group or for the conference resource itself (see resource-confmaxsize on page 133 and the Media Control Platform conference.limit parameter on page 165).


service-type The default type of service the IVR Profile provides.The default service-type does not preclude the use of other service types within the application as well.

• ccxml

• conference

• voicexml

Default value: voicexml

sip.sessiontimer The timeout value, in seconds, for the SIP session that executes for this IVR Profile. If the Resource Manager receives no SIP messages associated with this call leg within the timeout interval, the Resource Manager considers the call leg to have ended.For the call leg associated with this IVR Profile, the value of this sip.sessiontimer parameter overrides session expiry timeouts that are set at the level of the tenant, the Resource Manager, and the resource.


User’s Guide 143


gvp.policy SectionTable 23 describes the parameters in the gvp.policy section. These parameters enable you to configure policies for the Resource Manager—for example, to specify which requests the Resource Manager will allow, or to attach certain Request-URI parameters to send to the endpoint to enable or disable particular features.

Table 22: IVR Profile Configuration Options—gvp.log Section


metricsfilter The filter that determines which metrics in the Data Collection Sink (DATAC) will be forwarded to the Reporting Server.If this parameter is set, the value will override the default DATAC filter for the component (see ems.dc.default.metricsfilter on page 105), for sessions that execute under this IVR Profile.The Resource Manager passes this property value to the component in a SIP custom header.




The wildcard character (*) means all.Default value: Empty

Table 23: IVR Profile Configuration Options—gvp.policy Section


conference-allowed Specifies whether a Resource Manager session is allowed to use a conference.

• True

• False

Default value: True

dialing-rule-forbidden-respcode

The SIP response code that is sent in the SIP response when a call is rejected because of a dialing rule (gvp.policy.dialing-rules.rule-<n>).

• <sipcode>;<desc>

• <sipcode>

where: • <sipcode> is an integer in

the range of 400–699.• <desc> is any string.Default value: 403



dialing-rule-forbidden-set-alarm

Specifies whether an alarm will be raised for the corresponding policy violation.

• True

• False


external-sip-allowed Specifies whether a Resource Manager session is allowed to use an external SIP service.

• True

• False

Default value: True

inbound-usage-limit The number of times this IVR Profile may be concurrently in use for an inbound call.


mcp-asr-usage-mode

Specifies whether there will be one Voice Resource Management (VRM) session for the entire call, or whether a separate VRM session will be opened for each recognition request.A single session for the entire call (mcp-asr-usage-mode = per-call) means that each call may have multiple recognition sessions.If this parameter is set not to enable a single session for the entire call (mcp-asr-usage-mode = per-utterance), each VRM session is closed when the recognition request completes, either successfully or unsuccessfully (such as no match). Therefore, each call may have multiple VRM sessions.The Resource Manager passes this value to the Media Control Platform in a Request-URI parameter. The value of this parameter overrides a similar parameter that is set for the Media Control Platform overall (asr.load_once_per_call), if the settings are not consistent. See the description of the asr.load_once_per_call parameter on page 164 for more information about the implications of this setting.

• per-call

• per-utterance

Default value: per-call

mcp-max-log-level The Maximum Log Level allowed for a Media Control Platform application. The Resource Manager passes this value to the Media Control Platform in a Request-URI parameter.

An alphanumeric string.Default value: Empty

Table 23: IVR Profile Configuration Options—gvp.policy Section (Continued)


User’s Guide 145


mcp-sendrecv-enabled

Specifies whether a Media Control Platform is allowed to perform <send> and <receive> extensions. The Resource Manager passes this value to the Media Control Platform in a Request-URI parameter.

• True

• False

Default value: True

outbound-call-allowed

Specifies whether a Resource Manager session is allowed to make an outbound call. An outbound call in this context is a call through a gateway or to a SIP device not managed by the Resource Manager.

• True

• False

Default value: True

outbound-usage-limit

The number of times this IVR Profile may be concurrently in use for an outbound call.


<service>-capability-requirement

A list of name-value pairs that specify the capabilities that are required when the specified <service> is invoked in the context of this IVR Profile.Valid values for <service> are:• ccxml

• conference

• voicexml

Items in the name-value pair list are separated by a semi-colon. The value side of each name-value pair can itself be a comma-separated list of capabilities. Each set of values must be unique.The Resource Manager will direct interactions to a resource group only if the resource group capabilities exactly match the capability requirements specified in this option (see the <Logical Group>.capability option in Table 20 on page 131).

<cap_NameA>=<cap_ValueA> [;<cap_NameB>=<cap_ ValueB>;...]

where:• <cap_NameX> is the name

of the capability.• <cap_ValueX> is a

comma-separated list of values.

Example:lang=en-US;grammar=grxml,gsl






<service>-forbidden-respcode

The SIP response code that is sent in the SIP response when a request for the service is rejected because the service is not allowed in the session (gvp.policy.<service>-allowed=false).Valid values for <service> are:• conference

• external-sip

• outbound-call

• transfer

• voicexml-dialog


• <sipcode>



<service>-forbidden-set-alarm

Specifies whether an alarm will be raised for the corresponding policy violation. Valid values for <service> are:• ccxml

• conference

• external-sip

• outbound-call

• transfer

• voicexml-dialog

• True

• False


<service>-usage-limit

The number of times the specified service may be invoked in the context of this IVR Profile. Valid values for <service> are:• ccxml

• conference

• voicexml


<service>-usage-limit-exceeded-respcode

The SIP response code that is sent in the SIP response when a request for a service is rejected because the usage limits for that service (gvp.policy.<service>-usage-limit or gvp.policy.<service>-usage-limit-per-session) have been reached. Valid values for <service> are:• ccxml

• conference

• voicexml


• <sipcode>





User’s Guide 147


<service>-usage-limit-exceeded-set-alarm

Specifies whether an alarm will be raised for the corresponding policy violation. Valid values for <service> are:• ccxml

• conference

• external-sip

• inbound

• outbound

• voicexml

• True

• False


<service>-usage-limit-per-session

The number of times the specified service may be invoked in the context of this instance of a Resource Manager session. Valid values for <service> are:• ccxml

• conference

• voicexml


transfer-allowed Specifies whether a Resource Manager session is allowed to perform a transfer by using a SIP REFER request within the existing SIP session.

• True

• False

Default value: True

usage-limit-exceeded-respcode

The SIP response code that is sent in the SIP response when a call is rejected because the usage limits specified in the following configuration options have been reached:• gvp.policy.usage-limits

• gvp.policy.outbound-usage-limit

• gvp.policy.inbound-usage-limit


• <sipcode>


the range of 400–699.• <desc> is any string.Default value: 480 (Temporarily unavailable)

usage-limit-exceeded-set-alarm

Specifies whether an alarm will be raised for the corresponding policy violation.

• True

• False


usage-limits The number of times this IVR Profile may be concurrently in use.If a value is not specified (the default), there are no usage limits for the IVR Profile.






gvp.policy.dialing-rulesTable 24 describes the parameters in the gvp.policy.dialing-rules subsection.

use-same-gateway (For gateway service only) Specifies whether outbound calls to a gateway must use the same gateway that the Resource Manager session is currently using.A gateway resource becomes associated with a Resource Manager session when (a) the Resource Manager session is not already associated with another gateway resource and (b) one of the following occurs:• The Resource Manager receives a request from

a gateway resource.• The Resource Manager receives a request for a

gateway service and allocates it in accordance with the load-balancing scheme for the group.

• always—The Resource Manager must forward the request to exactly the same gateway resource already associated with the Resource Manager session, or else the request fails.

• preferred—The Resource Manager first tries to forward the request to the gateway resource already associated with the Resource Manager session, but tries other gateways if the first request fails.

• indifferent—The Resource Manager chooses a gateway in accordance with load-balancing scheme for the group.

Default value: always

voicexml-dialog-allowed

Specifies whether a Resource Manager session is allowed to use a VoiceXML service.

• True

• False

Default value: True



User’s Guide 149


gvp.service-parameters SectionTable 25 describes the parameters in the gvp.service-parameters section. The Resource Manager uses these values to add, modify, or delete Request-URI parameters in the SIP requests that it forwards.

Table 24: IVR Profile Configuration Options—gvp.policy.dialing-rules Section


rule-<n> For each <n>, this parameter specifies a dialing rule that the Resource Manager will use to determine if an address towards a gateway is allowed.<n> is a positive integer in the range of 1–10000.The rules are applied in rule number order.Example:To reject outbound calls to 911, allow calls to toll-free numbers, and reject calls to long-distance numbers, specify the following set of rules:gvp.policy.dialing-rules.rule-1: r,911

gvp.policy.dialing-rules.rule-2: a,1800*




gvp.policy.dialing-rules.rule-6: r,1*

<rule-type>;<regex>

where:• <rule-type> is either a or r, where a = allow and r = reject.

• <regex> is a regular expression.




gvp.service-prerequisite SectionTable 26 describes the parameters in the gvp.service-prerequisite section. These parameters provide information that the Resource Manager needs when it uses a default IVR Profile.

Table 25: IVR Profile Configuration Options—gvp.service-parameters Section


<service>.<param-name>

Each parameter that you create in this section takes the form of a pair of strings that determine whether a Request-URI parameter called <param-name>, with a value specified in <value>, will be included in forwarded SIP requests.Valid values for <service> are:• ccxml

• conference

• external

• gateway

• voicexml

The Resource Manager will apply this parameter to a SIP request only if the specified <service> is invoked by the SIP request.• Setting the <value-type> to undefined deletes

the <param-name> parameter from the incoming SIP request.

• Setting the <value-type> to fixed overrides the <param-name> parameter value in the incoming SIP request.

• Setting the <value-type> to default provides a default value for the <param-name> parameter in the outgoing SIP request, if the <param-name> parameter does not already exist.

<value-type>, <value>

where <value> is any string and <value-type> is:• undefined—The SIP Request-URI parameter with the specified <param-name> will not be in the forwarded request (even if the parameter was already in the incoming request).

• fixed—The parameter <param-name>=<value> will be in the SIP Request-URI.

• default—If the SIP Request-URI parameter with name <param-name> already exists, it will be left unmodified in the SIP Request-URI, but if the incoming request does not already include the parameter, the parameter <param-name>=<value> will be added to the SIP Request-URI.

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Chapter 6: Provisioning IVR Profiles Mapping IVR Profiles to Dialed Numbers

Mapping IVR Profiles to Dialed NumbersDialed numbers are the DNs obtained from Dialed Number Identification Service (DNIS). The Resource Manager can be configured so that it obtains DNIS information from SIP Server (see rm.sip-header-for-dnis on page 126). • If your GVP configuration includes a mapping of IVR Profiles to DNs, the

Resource Manager will use the DNIS information to determine which IVR Profile to invoke for the session.

Table 26: IVR Profile Configuration Options—gvp.service-prerequisite Section


alternatevoicexml (For voicexml service only) The URL to an alternative initial page that the Media Control Platform will use if the request to the initial-page-url fails.Before it forwards the service request, the Resource Manager inserts this information as the value of the gvp.alternatevoicexml SIP parameter.

Any valid URL.Default value: Empty

conference-id (Mandatory for conference service) The conference identifier.Before it forwards the service request, the Resource Manager replaces the user part of the SIP Request-URI with conf=<conference-id>.The Resource Manager uses the conference-id to ensure that it routes all requests for the same conference to the same conference resource, even if the requests originate from different Resource Manager sessions.

Any alphanumeric string, without spaces.Default value: Empty

default-properties-page

(For voicexml service only) The URL to a page containing the default properties and handlers.Before it forwards the service request, the Resource Manager inserts this information as the value of the gvp.defaultsvxml SIP parameter.


initial-page-url (Mandatory for voicexml and ccxml services) The URL of the initial page to be invoked.Before it forwards the service request, the Resource Manager inserts this information as the value of the voicexml or ccxml SIP parameter.



Chapter 6: Provisioning IVR Profiles Mapping IVR Profiles to Dialed Numbers

• If you do not map IVR Profiles to DNs, the Resource Manager will use the default IVR Profile that you specify for the tenant (see gvp.general.default-application on page 154).

There are two ways to map IVR Profiles to DNs:• Create rules on the Provisioning > Voice Platform > Dialed Number

Mapping tab. All IVR Profiles in the GVP environment are available for selection.

• Create rules for a particular IVR Profile on the Provisioning > Voice Platform > IVR Profile > <IVR Profile> > Number Mapping tab.

Whichever method you use, the mapping is updated in both contexts, as well as on the Provisioning > Environment > Tenants > <tenant> > Options > Annex list tab (gvp.dnis-range.<range> parameters).The following procedure describes how to create mapping rules on the Voice Platform tab.

Procedure:Mapping IVR Profiles to DNs

Purpose: To associate IVR Profiles with DNs so that the Resource Manager can use DNIS information to invoke the required GVP services.

Prerequisites

• The IVR Profiles have been created. For more information about creating an IVR Profile, see the chapter about post-installation activities in the Genesys Voice Platform 8 Deployment Guide. For more information about configuring the IVR Profile, see “IVR Profile Configuration Options” on page 141.


Start of procedure

1. Go to the Provisioning > Voice Platform > IVR Profile > <IVR Profile> > Number Mapping tab.

2. In the menu bar of the tab, click New Rule.The Specify Dialed Number Mapping Rule dialog box displays (see Figure 8).

User’s Guide 153

Chapter 6: Provisioning IVR Profiles Specifying Tenant Environment Settings

Figure 8: Specifying a Dialed Number Mapping Rule

3. In the Minimum and Maximum text boxes, enter the start and end numbers of the DN range. For a single DN, rather than a range, enter the same number in both text boxes.

4. From the IVR Profile drop-down list, select the required IVR Profile.5. Click OK.

The list of mapping rules on the number mapping tab is updated to include the new rule. The display also includes a column (State) and icon that indicate the status of the rule—valid or invalid.A range is valid if each DN in the range is not already specified as part of another mapping.

6. If you were specifying the rule on the IVR Profile Number Mapping tab, click Apply, then Save.

End of procedure

Specifying Tenant Environment SettingsThe Tenant configuration options determine how the Resource Manager will use IVR Profiles in the tenant’s environment.



Setting options at the tenant level sets values that are inherited as defaults by the IVR Profiles. You can override these settings for individual IVR Profiles by setting different values for the equivalent options in the IVR Profile.Configure the Environment tenant parameters in the Genesys Administrator on the Provisioning > Environment > Tenants > Environment [tenant] > Options > Annex list tab.All changes to these parameters take effect with the next session that uses the IVR Profile.The following tables describe the Tenant configuration options:• gvp.dnis-range Section• gvp.general Section—Table 27• gvp.policy Section—see page 155 and Table 23 on page 143• gvp.service-parameters Section—see page 155 and Table 25 on page 150

gvp.dnis-range SectionThe parameter list in this section shows the mapping of dialed numbers to IVR Profiles. The list is automatically updated when you use one of the methods to map a DNIS range to an IVR Profile (see “Mapping IVR Profiles to Dialed Numbers” on page 151).Do not create or modify the mapping from this section.

gvp.general SectionTable 27 describes the parameters in the gvp.general section. These parameters specify general configuration information for the Resource Manager in the tenant’s environment.

Table 27: Tenant Configuration Options—gvp.general Section


default-application (Mandatory) The default IVR Profile for a request to the Resource Manager. The Resource Manager uses the default IVR Profile if the incoming request does not contain information to map the request to an application.

<IVR Profile>

where <IVR Profile> is the name of the IVR Profile that you assigned when you created the IVR Profile object.Default value: Empty

User’s Guide 155


gvp.policy SectionThe parameters in this section are identical to the configuration parameters in the gvp.policy section of the IVR Profile object. These parameters enable you to configure policies for the Resource Manager—for example, to specify which requests the Resource Manager will allow, or to attach certain Request-URI parameters to send to the endpoint to enable or disable particular features.For more information about the configuration options in the gvp.policy section, see Table 23 on page 143.

gvp.service-parameters SectionThe parameters in this section are identical to the configuration parameters in the gvp.service-parameters section of the IVR Profile object. The Resource Manager uses these values to add, modify, or delete Request-URI parameters in the SIP requests that it forwards. For more information about the configuration options in the gvp.service-parameters section, see Table 25 on page 150.

sip.sessiontimer The timeout value, in seconds, for the SIP session that executes for this IVR Profile. If the Resource Manager receives no SIP messages associated with this call leg within the timeout interval, the Resource Manager considers the call leg to have ended.For the call leg associated with this IVR Profile, the value of this sip.sessiontimer parameter overrides session expiry timeouts that are set at the level of the Resource Manager, but may be overridden by the sip.sessiontimer setting for the IVR Profile.For more information about how the Resource Manager uses expiry timeouts to manage sessions, see “Session Management” on page 27.


usage-limits The number of times a Resource Manager session can be concurrently in use in the context of any IVR Profile.


Table 27: Tenant Configuration Options—gvp.general Section (Continued)




User’s Guide 157

Chapter

7 Configuring the Media Control PlatformThe Media Control Platform is the Genesys Voice Platform (GVP) component that provides media-centric services. This chapter provides information about configuring the Media Control Platform and, if required, provisioning the resources for Automatic Speech Recognition (ASR) and Text-to-Speech (TTS).This chapter contains the following sections:

Task Summary: Configuring the Media Control Platform, page 157Enabling ASR and TTS, page 159Important Media Control Platform Configuration Options, page 162Important MRCP Server Configuration Options, page 178

Task Summary: Configuring the Media Control Platform

Table 28 summarizes the configuration steps and options to implement Media Control Platform functionality in your GVP deployment.


Chapter 7: Configuring the Media Control Platform Task Summary: Configuring the Media Control Platform

Table 28: Configuring the Media Control Platform


Integrate the Media Control Platform with the Resource Manager.

Point the Media Control Platform to the Resource Manager as the SIP Proxy server, and define the properties for SIP communications. Key configuration options are:• sip.transport.x (see page 93)• sip.routeset or sip.securerouteset (see page 90)To secure SIP communications, ensure that you specify a transport for the Transport Layer Security (TLS) protocol and a secure routeset for outbound calls.For additional, relevant configuration options, see “Configuring SIP Communications and Routing” on page 86.

(Optional) Secure the media channel between the Media Control Platform and the remote endpoint.

1. Enable Secure Real-time Transport Protocol (SRTP) by specifying the required mode (accept-only or offer) in the mpc.srtp.mode parameter (see page 166). By default, SRTP is not enabled.

2. If necessary, modify the default values for the encryption and authentication algorithms (the cryptographic suites) and session parameters that the Media Control Platform will advertise in the SDP crypto attribute:mpc.srtp.cryptomethodsmpc.sessionparamsmpc.sessionparamsoffer

For more information about secure communications in GVP, see “Secure Communications” on page 70.

If required for your deployment, provision the third-party Media Resource Control Protocol (MRCP) servers for ASR and TTS.

See “Enabling ASR and TTS” on page 159.



Configure logging. See “Configuring Logging” on page 111.

User’s Guide 159

Chapter 7: Configuring the Media Control Platform Enabling ASR and TTS

Enabling ASR and TTSThe following procedure describes how to create and configure MRCP server Applications, to provision ASR and TTS speech resources for the GVP deployment.

Tune Media Control Platform performance.

• Configure appropriate maximums and timeouts for your deployment. Consider the following options, in particular:vxmli.cache.document.max_count (default is 50)vxmli.cache.document.max_size (default is 1000000 bytes)vxmli.max_num_documents (default is 2000)vxmli.initial_request_fetchtimeout (default is 30000 ms)vxmli.max_num_sessions (default is 10000)

• If your deployment includes ASR and TTS, consider the following options, which affect the MRCP Client behavior:vrm.client.timeout (default is 10000 ms)stack.stack.connection.timeout (default is 10000 ms)

• See also “Configuring Session Timers and Timeouts” on page 117.

• It is usually not necessary to modify the default settings for the media processing behavior of the Media Server (mpc and mtinternal configuration sections). However, review the mediamgr.* options in the mcp configuration section, to verify that they are optimal for your deployment.

Customize Media Control Platform behavior in relation to VoiceXML applications.

• Review and, if necessary, modify the configuration options in the vxmli configuration section (see the Genesys Voice Platform 8.0 Configuration Options Reference). Some of the important vxmli options are described in Table 29 on page 164.

• Consider also the parameters in the sip configuration section that specify what parts of SIP messages are exposed to the VoiceXML application (for example, in.invite.headers and in.invite.parameters). For the list of SIP headers that are known to GVP, see Table 71 on page 295.


See “Configuring Session Timers and Timeouts” on page 117.

Customize Media Control Platform messaging.


Table 28: Configuring the Media Control Platform (Continued)




Procedure:Provisioning ASR and TTS resources

Purpose: To provide an overview of the steps to configure logical GVP MRCPv1 or MRCPv2 speech server Applications, to provide a presence for third-party speech engines in the Genesys Configuration Layer.

Repeat this procedure as required to create the necessary Application objects. You must create a separate Application for each third-party speech server in your deployment. The Application type is Third Party Server.

Prerequisites


• The Media Control Platform Installation Package (IP) is available.

Start of procedure

1. Create the MRCPv1 or MRCPv2 Application object.a. Import the required Application Template from the Media Control

Platform Installation Package (IP).The following Application Templates are available:

b. Import metadata into the Application Template.c. On the Provisioning > Environment > Applications tab, create and

name the new Third Party Server Application, based on the required Application Template.

For detailed information about importing Application Templates and metadata, and creating Applications from the templates, see the appendix about pre-installation activities in the Genesys Voice Platform 8.0 Deployment Guide.

2. On the Provisioning > Environment > Applications > <MRCP Server> > Settings tab, configure the options in the provision configuration section, as required for your deployment.a. At a minimum, you must specify values for the following options:

• vrm.client.resource.address—The IP address of the speech resource.

• vrm.client.resource.uri—The URI to the speech resource.

• MRCPv1_ASR • MRCPv2_ASR

• MRCPv1_TTS • MRCPv2_TTS

User’s Guide 161


• vrm.client.resource.port—The port on the speech resource for communication with the Media Control Platform MRCP Client. The default value is 4900 for MRCPv1, and 5060 for MRCPv2.

b. If necessary, modify the vrm.client.resource.name parameter to add the name of your speech resource vendor. Modify this parameter:• For ASR, if your vendor is not SPEECHWORKS. Continue at

Step 3.• For TTS, if your vendor is not REALSPEAK. Continue at Step 4.

For other important configuration options that you may need to modify, see “Important MRCP Server Configuration Options” on page 178.

3. If you added an ASR speech vendor (Step 2b), you must also update the hotkey grammar directory and path.a. On the Provisioning > Environment > Applications > <MRCP Server>

> Settings tab, modify the following parameters to use the name of your ASR vendor (vrm.client.resource.name):• vrm.client.HotKeyBasePath=/vggrammarbase/<vendor

name>/hotkey

• vrm.client.HotKeyLocalPath=$InstallationRoot$/grammar/<vendor n

ame>/hotkey

For more information about these parameters, see page 181.b. Click Save or Apply to save the MRCP server configuration.c. On the Media Control Platform host, go to the grammar directory in the

path specified in vrm.client.HotKeyLocalPath.d. Create a new directory with the <vendor name> specified in

vrm.client.resource.name.e. In the <vendor name> directory, create a new directory called hotkey.f. Copy the hotkey grammar file from

$InstallationRoot$\grammar\<vendor name>/hspeechworks\hotkey to the new hotkey directory.

4. If required, configure the TTS vendor-specific parameters that will be sent in SET-PARAM requests:a. In the provision section on the Provisioning > Environment >

Applications > <MRCP Server> > Options tab, add a new parameter, vrm.client.TTSVendorSpecific.xxxxxx.

This defines one arbitrary TTS vendor-specific parameter to be sent to the MRCP server.

b. Define as many vendor-specific keys as you require for the desired vendor-specific key-value pairs, using the following format:vrm.client.TTSVendorSpecific.param<n>=value<n>

c. Click Save or Apply to save the MRCP server configuration.


Chapter 7: Configuring the Media Control Platform Important Media Control Platform Configuration Options

5. On the Provisioning > Environment > Applications > <Media Control Platform> > Configuration tab, create the connection between the Media Control Platform and the MRCP server. For more details, see the procedure to assign the MRCP server to the Media Control Platform, in the chapter about post-installation activities in the Genesys Voice Platform 8.0 Deployment Guide.

End of procedure

Important Media Control Platform Configuration Options

This section describes the key configuration options that you either must or may want to customize.Configure the options on the Genesys Administrator on the Provisioning >

Environment > Applications > <Media Control Platform> > Settings tab. For the detailed steps to configure option settings, see Viewing or modifying GVP configuration parameters, page 78.The configurable Media Control Platform parameters are in the following configuration sections:• asr

• calllog

• conference—Parameters determine the default behavior of the Conference application module, for NETANN-initiated conference calls.

• email—Parameters enable you to configure e-mail address information for maintainer e-mails.

• ems (see Table 13 on page 104)—Parameters determine EMS Reporting behavior for call detail records (CDRs) and metrics.

• log (see “Configuring Logging” on page 111)—Parameters determine behavior for Management Framework logging.

• mpc and mtmpc—Parameters determine the default media processing and transport behavior of the Media Processing Component (MPC), or Media Server.

• mtinternal—Parameters determine the behavior of the Internal Media Transport application module, which is responsible for managing internal media transmission between the Media Server and the ASR and TTS speech engines. This internal data transmission uses RTP.

• sessmgr—Parameters determine call control and platform-level behavior of the Call Manager API (CMAPI) application modules that are loaded at startup.

User’s Guide 163


Note: Genesys recommends that you do not modify the default values, unless you are an advanced user who needs to use special CMAPI applications for your deployment.

• sip—Parameters integrate the Media Control Platform with the SIP Proxy (the Resource Manager). These parameters determine the behavior of the SIP Line Manager application module, and configure the supported transport interfaces.

• stack—Parameters relate to the MRCP stack and determine the way the Media Control Platform manages connections to the external MRCP server.

• vrm—Parameters determine the behavior of the MRCP Client. These parameters relate to the Voice Resource Management (VRM), or Speech Resource Management (SRM), module.

• vxmli—Parameters determine the behavior of the Next Generation Interpreter (NGI).

Table 29 provides information about important Media Control Platform parameters that are not described in Chapter 4 on page 85. Table 29 provides parameter descriptions as well as the default parameter values that are preconfigured in the Media Control Platform Application object.Unless indicated otherwise, all changes take effect on restart.For information about all the available configuration options for the Media Control Platform, see the Genesys Voice Platform 8.0 Configuration Options Reference.



Table 29: Selected Media Control Platform Configuration Options


asr Section

load_once_per_call Specifies whether there will be one VRM session for the entire call, or whether a separate VRM session will be opened for each recognition request.A single session for the entire call (load_once_per_call = 1) means that each call may have multiple recognition sessions.If this parameter is set not to enable a single session for the entire call (load_once_per_call = 0), each VRM session is closed when the recognition request completes, either successfully or unsuccessfully (such as no match). Therefore, each call may have multiple VRM sessions.Having multiple VRM sessions in a call may improve the efficiency of ASR server license usage. However, be aware of the following possible consequences:• There will be longer delays on speech barge-

in.• Some recognizer servers delete saved

utterance data after each VRM session. In these cases, the VoiceXML application cannot refer to the saved utterance file after the recognition session.


• 0—Single session not enabled.

• 1—Single session enabled.

Default value: 1 (only one VRM session for the entire call)

conference Section

audio_format The audio codec for conference. • pcmu

• pcma

• g726-16

• g726-24

• g726-32

• g726-40

• l16

• gsm

Default value: pcmu

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limit The maximum number of participants allowed for a conference initiated by the conferencing application.

An integer in the range 2–32.Default value: 32

mpc Section

appendrejcodec Specifies whether GVP will advertise all supported codecs when it generates a Session Description Protocol (SDP) answer or SDP offer.Even if codecs are rejected or not presented in the caller’s SDP message, the platform will still support receiving these codecs. The platform will not send for the SDPs unless a payload is presented by the caller.Changes take effect: Immediately.

• 0—GVP will not advertise all supported codecs.

• 1—GVP will advertise all supported codecs.

Default value: 0

codec A space-separated list of the codecs that correspond to the platform capabilities advertised with SDP.The list controls which codecs the Media Control Platform offers to the remote party, for media sent from the remote party to GVP.Changes take effect: Immediately.

• amr

• g726

• g729

• gsm

• h263

• h263-1998

• pcmu

• pcma

• telephone-event

• tfci

Default value: pcmu pcma g726 gsm h263 h263-1998 telephone-event

Table 29: Selected Media Control Platform Configuration Options (Continued)




codecpref Specifies whether remote or local preferences will be used to interpret the list of accepted codecs. • Local preferences means that the effective

accept list is the locally configured accept list, filtered to include only those capabilities also offered by the remote entity.

• Remote preferences means that the effective accept list is the list of formats offered by the remote entity, filtered to include only those entries also on the locally configured list.


• l—Local preferences will be used.

• r—Remote preferences will be used.

Default value: r

default_audio_format The default audio format for the Call Manager. • ALAW

• ULAW

Default value: ULAW

srtp.mode The mode of operation with regard to Secure Real-Time Transport Protocol (SRTP).For offer mode:• If the other side ignores SRTP, the platform

will fall back to non-SRTP mode.• If a previously negotiated m-line is used in a

reoffer or if the far end requests an offer, and that m-line did not have SRTP negotiated, SRTP will not be added.

• If the far end reoffers and adds SRTP to a previously negotiated m-line, SRTP will be negotiated.

• none—No SRTP support. The Media Control Platform will ignore the crypto attribute in SDP offers.

• accept_only—SRTP is supported for SDP offers sent to the Media Control Platform, but the platform will not add SRTP to m-lines in outgoing offers that did not previously contain it.

• offer—SRTP is supported for SDP offers sent to the Media Control Platform, and will be included in all outgoing SDP offers.

Default value: none



User’s Guide 167


sip Section

(Note: For additional important options in this configuration section, see also “Configuring SIP Communications and Routing” on page 86.)

confserver (Mandatory if a conferencing server is deployed in the network) The address of the GVP conferencing server.Note: Underscores are not valid characters in the IP address.

<IP address>[:<port>]

where:• <IP address> is the IP

address of the conferencing server host.

• <port> is the SIP port number. You must specify this value if the conferencing server uses a non-default SIP port (default is 5060).


defaultblindxfer The default transfer method for SIP, for blind transfers.For more information about the transfer types and methods, see “Transfers” on page 45.

• HKF—Hookflash• REFER—REFER-based

transfer• BRIDGE—Bridge-based

transfer• REFERJOIN—Consultative REFER transfer

• MEDIAREDIRECT—Media redirect transfer

Default value: REFER

defaultbridgexfer The default transfer method for SIP, for bridge-type transfers.For more information about the transfer types and methods, see “Transfers” on page 45.

• BRIDGE—Bridge-based transfer


Default value: BRIDGE





defaultconsultxfer The default transfer method for SIP, for consult-type transfers.For more information about the transfer types and methods, see “Transfers” on page 45.

• HKF—Hookflash• BRIDGE—Bridge-based



Default value: REFERJOIN

defaultgw The default gateway host and port that will be used for SIP calls (transfer, call, or remote dial) to a telephone, if the destination address does not specify a gateway.If this parameter is not specified, telephony calls that do not specify a gateway in the destination address will fail.Example:If sip.defaultgw=pstn-gw.voiceplatform.com:5060 and a SIP call is placed to telephone number 123456789, the SIP Line Manager translates the destination address to sip:123456789@default-gw, and the call is routed to port 5060 on host pstn-gw.voiceplatform.com.

<Host name or IP address>:<SIP port>


defaulthost The default host and port that the Media Control Platform will use for SIP calls (transfer, call, or remote dial), if the destination address does not contain a host name or IP address.If this parameter is not specified, calls that do not specify a host in the destination address will fail.Example:If sip.defaulthost=voiceplatform.com:5060 and a SIP call is placed to address sip:1234@, the destination address is translated to: sip:[email protected]:5060

<Host name or IP address>:<SIP port>




User’s Guide 169


deferoutalerting Enables early media for an outbound call, by specifying whether the CallOutAlerting response to the session manager will be deferred until the media session is initialized and registered.If enabled, the session manager can start performing media operations on the channel as soon as the session manager receives the CallOutAlerting notification.

• 0—CallOutAlerting will not be deferred.

• 1—CallOutAlerting will be deferred.

Default value: 0

dnis_correlationid_length

The length of the correlation ID, within the user-id portion of the DNIS. The correlation ID is the portion of the user-id that will be stripped, in order to isolate the DNIS.Note: In the special case where the correlation ID is all of the user-id, the ampersand character (@) will also be stripped away from the DNIS, because @<hostname> does not make sense.

A non-negative integer. Default value: 0 (no correlation ID)

dnis_correlationid_offset

The offset that specifies where the correlation ID starts, within the user-id portion of the DNIS. The correlation ID is the portion of the user-id that will be stripped, in order to isolate the DNIS.

Any integer. A negative value indicates that the offset is from the right.Default value: 0 (no offset)

enablesendrecvevents

Enables the sending and receiving of SIP INFO messages for VoiceXML application usage. This parameter does not affect SIP INFO messages used for other purposes (for example, DTMF).Changes take effect: Immediately.

• true—VoiceXML applications are enabled to send and receive SIP INFO messages.

• false—VoiceXML applications cannot send and receive SIP INFO messages.

Default value: true





hfdisctimer The timeout value, in milliseconds, to terminate a SIP hookflash transfer.• If sip.hftype=0 (wait for disconnection), the

transfer is treated as failed if a BYE is not received from the remote end before this timeout expires.

• If sip.hftype=1 (force disconnection), the transfer is always treated as successful. If a BYE is not received from the remote end before this timeout expires, then a BYE will be sent from the local end.

Any non-negative integer.Default value: 5000

hfprefix The SIP hookflash transfer dialing prefix.Examples:• sip.hfprefix=none means the dial string is

exactly as specified in the transfer.• sip.hfprefix=! means dial a hookflash.• sip.hfprefix=*8,, means dial *8 followed by

two pause durations.

A string containing one or more of the following characters: 0–9 , ! * noneDefault value: !

hfstopdial The digits to dial to stop a hookflash transfer. Dialing the digits specified in this parameter will abort a multi-phase hookflash. The connection is switched back to the original caller.

A string containing one or more of the following characters: 0–9 !Default value: !

hftype Specifies the type of hookflash transfer for SIP. • 0—Wait for disconnection.

• 1—Force disconnection.Default value: 0



User’s Guide 171


in.<SIP request>.headers

The list of header names from incoming <SIP request> messages that will be exposed to the VoiceXML application, where <SIP request> is one of:• BYE

• INFO

• INVITE

The names of the exposed headers appear in the application in the following format:sip.invite.<headername>=<value>

<Header1> [<Header2>...]

where <HeaderX> is:• A header name—Each

specified header name will be exposed.

• *—All header names will be exposed.

• none—No header names will be exposed. If any other value is specified alongside none, then none is ignored.

Example: From To ViaDefault values: • For BYE requests: Reason• For INFO and INVITE

requests: *

in.invite.params The list of header names from incoming INVITE requests whose parameters will be exposed to the VoiceXML application.The exposed parameter values appear in the application in the following format:sip.invite.<headernam>.<paramname>=<value>





Default value: RequestURI

info.contenttype The content type of outgoing SIP INFO messages that correspond to VoiceXML application <log> events.A VoiceXML application can trigger the sending of a SIP INFO message by using the <log> tag with dest="callmgr". Call Manager will then send a SIP INFO message to the remote end. The content of the SIP INFO message is the content of the <log> tag.

A string indicating the content type.Default value: application/text





localrtpaddr The Media Control Platform IP address to advertise for Real-time Transport Protocol (RTP).With multicast or proxied systems, you may need to specify what IP address to advertise in the SDP description for a session. By default, the IP address of the local system is retrieved by performing a standard gethostname(). However, if your system is multihomed or behind a firewall, use this parameter to control the IP address that is advertised.

<IP address>

Default value: Empty (which causes the local IP address to be determined automatically)

out.<SIP request>.headers

The list of header names from outgoing <SIP request> messages that will be exposed to the VoiceXML application, for customization. <SIP request> is one of:• INFO

• INVITE

• REFER

The customized names of the exposed headers appear in the application in the following format:sip.invite.<headername>=<value>




• *—All header names will be exposed.


Example: From To ViaDefault value: *

out.<SIP request>.params

The list of header names from outgoing <SIP request> messages whose parameters will be exposed to the VoiceXML application, for customization. <SIP request> is one of:• INVITE

• REFER

The exposed parameter values appear in the application in the following format:sip.invite.<headername>.<paramname>=<value>





Default value: RequestURI



User’s Guide 173


outcalluseoriggw Specifies how the Media Control Platform will determine which gateway to use for an outbound call or transfer, if the destination address does not contain a host name or IP address.Example:If sip.outcalluseoriggw=1 and the inbound call came from a gateway with host name 3000, the call will be placed to one of the following:• tel://3000

• sip:3000@—The ampersand character (@) is required to delimit the user part from the host part of the address.

• 0—The gateway specified in sip.defaultgw or sip.defaulthost will be used.

• 1—The gateway of the inbound call will be used.

Default value: 1

referxferhold Specifies whether to put the originating caller on hold (Invite hold) before the Media Control Platform sends the REFER message for a REFER or REFERJOIN transfer.

• 0—Original caller will not be put on hold.

• 1—Original caller will be put on hold.

Default value: 1





registration The settings for registering the Media Control Platform with the SIP Registrar.You can configure the system to register with one or more SIP registration servers on the network. To specify more than one registration entry, separate the entries with a pipe (|).The Media Control Platform will attempt to register with all defined registration entries, and will periodically reregister as required (in accordance with the <requested-expiry> parameter). The Media Control Platform will deregister when it shuts down.Example:ResourceManager.yourdomain.com:5064 [email protected] 60 - -|proxy2.yourdomain.com:5064 [email protected] 60 user password means that the Media Control Platform will register with the Resource Manager as SIP user [email protected], and with another SIP proxy as SIP user [email protected], with authentication user name user, and password password.

<registration-server> <register-as> <requested-expiry> <username> <password> [<routeset>]

where:• <registration-server>

is the host and port of the Resource Manager or other SIP registration server.

• <register-as> is the SIP identity of the Media Control Platform.

• <requested-expiry> is the duration of registration, in seconds.

• <username> is the user name when authentication is required by the server. This may or may not be the same as <register-as>. A dash (–) indicates that no user name is needed. If username=– (dash) and the server requests authentication, Anonymous is used.

• <Password> is the password associated with the authentication user name. To specify an empty string, use a dash (–).

• <Routeset> is a comma-separated list of the servers that the REGISTER messages will go through. If a route set is not defined, the REGISTER messages will be sent directly to the <registration-server>.




User’s Guide 175


sendalert The SIP response for alerting and intermediate provisional responses.Changes take effect: Immediately

• 0—No SIP response• 1—Send 180 RINGING

response• 2—Send 183 Session Progress response with SDP information

Default value: 1

sipinfoallowedcontenttypes

A space-delimited list of the content types that are allowed to be passed up to the VoiceXML application level in a SIP INFO message. Any content types that have not been defined will be ignored.

<Content type1>[<Content type2>...]

where <Content typeN> is:• An alphanumeric string—

Defines the content type.• An empty string—Allows

all content to be passed upstream.


transfermethods A space-separated list of the supported transfer methods for SIP.For more information about the transfer methods, see “Transfer Methods” on page 46.

• HKF—Hookflash• REFER—REFER-based



• none—No transfer methods for SIP

Default value: REFER REFERJOIN MEDIAREDIRECT





vxmlinvite Specifies whether VoiceXML URLs in SIP INVITE messages will be accepted, thereby bypassing the normal method of selecting a VoiceXML application on the basis of DNIS mapping.If vxmlinvite is enabled, the originator of a SIP call can specify the initial VoiceXML URL that will be fetched for the session. To implement this functionality, the originator of the SIP call must encode the Request-URI in the following special form:"sip:dialog.vxml.<URL>@host.com"

where the <URL> portion is encoded (for example, %3A).

• 0—VoiceXML URLs will not be accepted.

• 1—VoiceXML URLs will be accepted.

Default value: 1

warningheaders Specifies whether the Media Control Platform will send warning headers.Changes take effect: Immediately.

• 0—The Media Control Platform will send warning headers only when it receives an error response.

• 1—The Media Control Platform will always send warning headers, if there are any.

• 2—The Media Control Platform will never send warning headers.

Default value: 0

xfer.copyheaders A space-delimited list of the headers to be copied from inbound call INVITE requests to outbound call INVITE requests for the same VoiceXML session (in other words, for bridged and Release Link Transfer [RLT] calls).The headers are rescanned for the re-INVITE (the outbound call INVITE request), so changes that have been made to the values of the headers during the inbound call leg are applied on any outbound calls made within the call session.Changes take effect: Immediately.



specified header will be copied.

• *—All headers will be copied, including unknown headers.

• none—No headers will be copied.

Default value: *



User’s Guide 177


vrm Section

client.timeout The timeout interval, in milliseconds, for the MRCP client to wait for a response from the MRCP server.If no response is received within this timeout period, the request is deemed to have failed.

An integer in the range of 1–60000Default value: 10000

client.ping.frequency

The interval, in milliseconds, at which the MRCP Client pings each MRCP server that has been provisioned.The MRCP DESCRIBE method is used as a ping message.


client.ping.timeout The timeout interval, in milliseconds, for the MRCP client to wait for a ping response from the MRCP server.If no response is received within this timeout period, the MRCP server is considered to be unavailable. The MRCP Client disconnects from the server, and then periodically tries to re-establish a connection, at a retry interval specified in the client.ping.frequency parameter.Genesys recommends setting the client.ping.timeout value to twice the value of the client.ping.frequency parameter.


client.universals.uri The URI convention that the NGI uses to specify the universals grammars.

builtin:grammar/universals

Default value: builtin:grammar/universals

vxmli Section

conformance.strict_grammar_mode

Specifies whether the NGI will follow the VoiceXML specification strictly when handling the grammar element. The default value (false) means that the NGI will ignore the mode attribute for an external grammar.

• True

• False





Chapter 7: Configuring the Media Control Platform Important MRCP Server Configuration Options

Important MRCP Server Configuration Options

This section describes important configuration options that you either must or may want to customize.Configure the options in the Genesys Administrator on the Provisioning >

Environment > Applications > <MRCP Server> > Settings tab. For the detailed steps to configure option settings, see Viewing or modifying GVP configuration parameters, page 78.The configurable MRCP server options are in the provision configuration section. Table 29 provides information about these options. Table 29 provides parameter descriptions as well as the default parameter values that are preconfigured in the MRCPv1_ASR, MRCPv1_TTS, MRCPv2_ASR, and MRCPv2_TTS Application objects.All changes take effect on restart.For information about all the available configuration options for the MRCP servers, see the Genesys Voice Platform 8.0 Configuration Options Reference.

debug.enabled Enables real-time debugging for the platform. • True

• False


initial_request_method

The HTTP method to use for the initial request. • GET

• POST

Default value: GET

transfer.allowed Specifies whether dialog-initiated transfers are allowed.

• True

• False

Default value: True

userdata.prefix The string that, when used as a prefix in a SIP header, identifies userdata variables.

Any string.Default value: X-Genesys-



User’s Guide 179


Table 30: Selected MRCP Server Configuration Options


ASR and TTS

vrm.client.ConnectPerSetup

(For MRCPv2 only) Specifies whether the MRCP Client will create a new connection to the ASR or TTS server for each MRCP session setup.

• True

• False

Default value: True

vrm.client.resource.address

The IP address of the speech resource. <IP address>

Default value: 0.0.0.0

vrm.client.resource.name

The name of the speech resource vendor. <vendor_name>Default value:• For ASR: SPEECHWORKS• For TTS: REALSPEAK

vrm.client.resource.port

The port on the speech resource that will be used for communication with the Media Control Platform MRCP Client.

<port>

where <port> is an integer in the range of 1–60000.Default value:• For ASR servers: 5060• For TTS servers: 4900



vrm.client.resource.uri

The URI to the speech resource. • For MRCPv1 ASR: rtsp://<IP address>:<port>/media/speechrecognizer

• For MRCPv1 TTS: rtsp://<IP address>:<port>/media/speechsynthesizer

• For MRCPv2: sip:mresources@<IP address>:<port>

Default value:• For MRCPv1 ASR: rtsp://0.0.0.0:4900/media/speechrecognizer

• For MRCPv1 TTS: rtsp://0.0.0.0:4900/media/speechsynthesizer

• For MRCPv2: sip:[email protected]:5060

ASR Only

vrm.client.DisableHotWord

Specifies whether the platform will treat recognition-based barge-in as speech-based barge-in.Set this parameter to true for all ASR servers that do not support recognition-based barge-in.

• true—Recognition-based barge-in will be treated as speech-based.

• false—Recognition-based barge-in will not be treated as speech-based.

Default value: false

Table 30: Selected MRCP Server Configuration Options (Continued)


User’s Guide 181


vrm.client.HotKeyBasePath

The HTTP fetchable location for the hotkey grammars. The value of this parameter is concatenated with the IP address of the Media Control Platform to form a fetchable location for hotkey grammars.The <vendor name> in the path must be the same as the vendor name specified in vrm.client.resource.name on page 179.

/vggrammarbase/<vendor name>/hotkey

Default value: /vggrammarbase/speechworks/hotkey

vrm.client.HotKeyLocalPath

The local path for the hotkey grammars on the Media Control Platform. The MRCP Client uses the HotKeyBasePath to translates this address to the appropriate URI, which is sent to the ASR servers.

$InstallationRoot$/grammar/<vendor name>/hotkey

Default value: $InstallationRoot$/grammar/speechworks/hotkey

Table 30: Selected MRCP Server Configuration Options (Continued)




User’s Guide 183

Chapter

8 Configuring the Call Control PlatformThis chapter provides information about configuring the Call Control Platform and provisioning the device profiles in your Genesys Voice Platform (GVP) deployment.It contains the following sections:

Task Summary: Configuring the Call Control Platform, page 183Important Call Control Platform Configuration Options, page 185Configuring Device Profiles, page 188

Task Summary: Configuring the Call Control Platform

Table 31 summarizes the configuration steps and options to implement Call Control Platform functionality in your GVP deployment.


Chapter 8: Configuring the Call Control Platform Task Summary: Configuring the Call Control Platform

Table 31: Configuring the Call Control Platform


Integrate the Call Control Platform with the Resource Manager and Media Control Platform.

Point the Call Control Platform to the Resource Manager as the SIP Proxy server and interim target of media service requests, and define the properties for SIP communications. Key configuration options are:• mediacontroller.sipproxy (see page 188)• mediacontroller.bridge_server (see page 187)• sip.transport.x (see page 92)• sip.routeset or sip.securerouteset (see page 90)For additional, relevant configuration options and actions, see “Configuring SIP Communications and Routing” on page 86 and “Enabling Secure Communications” on page 93.

(Required only if you made TCP or TLS the preferred default transport protocol [see page 86])Ensure that the Request-URI header in SIP requests specifies the required transport protocol.

Modify the CCXML applications so that the Request-URI for any endpoints includes the transport=TCP or transport=TLS parameter.• Use CCXML hints in the <createcall>, <dialogprepare>, <dialogstart>, and <createconference> tags. For example:

<var name="hints" expr="new Object()"/>

<assign name="hints.requesturi" expr="new Object()"/>

<assign name="hints.requesturi.transport" expr="'tcp'"/>

<dialogstart src="'file:///C:\Program Files\GCTI\gvp\VP

Media Control Platform 8.0\MCP_80\helloaudio.vxml'"

hints="hints"/>

Ensure that the Call Control Platform can interact with all other SIP devices in your deployment.

Verify and, if necessary, modify the device profiles that have been provisioned. For more information, see “Configuring Device Profiles” on page 188.



Configure logging. See “Configuring Logging” on page 111.

User’s Guide 185

Chapter 8: Configuring the Call Control Platform Important Call Control Platform Configuration Options

Important Call Control Platform Configuration Options


Environment > Applications > <Call Control Platform> > Settings tab. For the detailed steps to configure option settings, see Viewing or modifying GVP configuration parameters, page 78.Except for some ems options, all changes to Call Control Platform options take effect immediately.The Call Control Platform configuration options are in the following configuration sections:• ccpccxml—Parameters determine the behavior of the Call Control Platform

in relation to the CCXML applications (for example, whether transfers through dialogs are allowed).

• ccxmli—Parameters determine the behavior of the CCXML Interpreter (for example, the HTTP port and URL for the IOProc function; maximums for the number of sessions, documents, and per-session conferences, connections, dialogs, processing threads, and so on).

Tune Call Control Platform performance.

Configure appropriate maximums and timeouts for your deployment. Consider the following options, in particular:• ccxmli.max_num_documents (default is 1000)• ccxmli.num_session_processing_threads (default is 5)• ccxmli.max_num_sessions (default is 1000)• ccxmli.max_conn_per_session (default is 100000)• ccxmli.max_dialog_per_session (default is 100)• ccxmli.max_conf_per_session (default is 100000)See also “Configuring Session Timers and Timeouts” on page 117.


See “Configuring Session Timers and Timeouts” on page 117.

Customize Call Control Platform messaging.


Table 31: Configuring the Call Control Platform (Continued)




• ems (see Table 13 on page 104)—Parameters determine EMS Reporting behavior for call detail records (CDRs) and metrics.

• log (see “Configuring Logging” on page 111)—Parameters determine behavior for Management Framework logging.

• mediacontroller and mediactrller—Parameters integrate the Call Control Platform, through the Resource Manager, with the Media Control Platform, which acts as a bridge server for call transfers and conferences.

• session—Parameters determine the behavior of the Call Control Platform during sessions (for example, whether unknown headers will be copied into forwarded SIP messages).

• sip—Parameters integrate the Call Control Platform with the SIP Proxy (the Resource Manager).

Table 32 provides information about important Call Control Platform parameters that are not described in Chapter 4 on page 85. Table 32 provides parameter descriptions as well as the default parameter values that are preconfigured in the Call Control Platform Application object.For information about all the available configuration options for the Call Control Platform, see the Genesys Voice Platform 8.0 Configuration Options Reference.

Table 32: Selected Call Control Platform Configuration Options


ccpccxml Section

default_uri The URI for the default CCXML application. <URI path to file>

Default value: file://$InstallationRoot$\config\default.ccxml

sip.allowedunknownheaders

A space-separated list of the unknown headers that can be sent in an outgoing SIP message.Genesys recommends that you allow the following headers:• Reason

• Warning

A string specifying a header name.Default value: Empty

User’s Guide 187


ccxmli Section

platform.save_<file type>_files

<file type> is either ccxml or script.Specifies whether fetch request, response, and data for each CCXML or ECMAScript file that is fetched and processed in a session will be saved to disk.This feature is convenient for debugging CCXML applications, particularly when CCXML pages are dynamically generated during a session.

• true

• falseDefault value: false

mediacontroller Section

bridge_server The Resource Manager IP address. The Call Control Platform sends requests to the Resource Manager to find a bridging server to use when two endpoints cannot be joined because of media bridging limitations (implicit conference and transcoding).The bridge server must be capable of:• Sending media to multiple endpoints.• Sending and receiving from distinct endpoints.• Performing transcoding.

<IP address>


bridge_server.profile The name of the device profile to use with the configured bridge server.For information about configuring device profiles, see “Configuring Device Profiles” on page 188.

<Device profile name>

Default value: Default Conference

Table 32: Selected Call Control Platform Configuration Options (Continued)



Chapter 8: Configuring the Call Control Platform Configuring Device Profiles

Configuring Device ProfilesThe Call Control Platform uses device profiles to determine the behavior of the devices with which it interacts, in order to produce the most appropriate SIP messages. (For more information about how the Call Control Platform uses device profiles, see “Device Profiles” on page 53.)

full_<media type>_codec

A space-separated list of the <media type> codecs that get set in the SDP in an initial offer when there is no media bridge. In other words, the media line that will be used to create a connectionless SDP.<media type> is either audio or video.For the codecs that the Media Control Platform and, therefore, the Call Control Platform support, see “Codec Negotiation” on page 43.

<payload>|<codec>|<MIME-type>|<rate>|<number of channels>

Default values:• Audio—

0|pcmu|audio/basic|8000|1

8|pcma|audio/x-alaw-

basic|8000|1 4|g723|none|

8000|1 18|g729|audio/

g729|8000|1 12|qcelp|

audio/QCELP|

8000|1 111|aurora|none|

8000|1 3|gsm|audio/x-

gsm|8000|1 105|AMR|audio/

AMR|8000|1 101|telephone-

event|none|8000|1

• Video—34|h263|video/H263|90000|

1 98|H264|video/H264|

90000|1

inbound_allowed_media

The default allowed media types for an inbound call. All inbound calls will be limited to this set of media types in terms of SDP exchange.

• audio

• audio video

• video

Default value: audio

sipproxy The Resource Manager address-of-record (AOR). The Resource Manager is the SIP Proxy that the Call Control Platform uses for outbound SIP requests.

<Resource Manager IP address>:<SIP port>


sip Section

See Table 9 on page 88.

Table 32: Selected Call Control Platform Configuration Options (Continued)


User’s Guide 189


Device Profile Configuration FileDevice profiles are defined in the <Call Control Platform Installation Directory>\config\ccpccxml_provision.dat file, which is installed when you install the Call Control Platform.For the format and syntax of device profile entries in the configuration file, see Provisioning Device Profiles for the Call Control Platform, page 194.For information about the properties that have been defined for the CCXML Device class of profiles, see Table 33.For information about the values that have been predefined for the default CCXML Device class of profiles, see Appendix D, “Default Device Profiles,” on page 304.

CCXML Device Class

Table 33 describes the properties that define the CCXML Device class of device profiles.

Table 33: CCXML Device Profile Class Properties

Property Description Valid Values

connectionless-sdp-type

Support for Connectionless SDP. Indicates the mechanism that should be used to indicate null SDP (in other words, no media streams) to the device.

• hold—Use hold SDP (for example, c=0.0.0.0).• non-routable—Use SDP with a non-routable

connection (for example, c=1.1.1.1).• none—The device does not support

connectionless SDP.

distinct-send-recv-support

Support for send-recv media lines.Indicates if the offer-answer user agent supports a sendonly media line and a recvonly media line on separate connections.

• true—The CCXML Device supports a sendonly media line and a recvonly media line on separate connections.

• false—The CCXML Device does not support a sendonly media line and a recvonly media line on separate connections.

multiple-recvonly-support

Support for multiple recvonly media lines.Indicates if the offer-answer user agent supports receiving multiple recvonly media lines.

• true—The CCXML Device supports receiving the recvonly attribute in multiple SDP media lines.

• false—The CCXML Device does not support receiving the recvonly attribute in multiple SDP media lines.



nomedia-SDP-support

Support for receiving SDP containing no media lines.Indicates if the offer-answer user agent supports receiving an SDP containing no media lines. The device answers with an SDP with no media lines and remains in a state with no media connections until a re-INVITE.

• true—The CCXML Device supports receiving SDP without media lines.

• false—The CCXML Device does not support receiving SDP without media lines.

offer-answer-support

Support for the Offer-Answer model.Indicates if the device supports the offer-answer model described in RFC 3264 (in other words, whether the device will respond to SDP offers with an answer according to the rules defined in the RFC).

• true—The CCXML Device supports the offer-answer model.

• false—The CCXML Device does not support the offer-answer model.

offer-less-invite-support

Support for INVITE requests that do not contain an SDP offer.Indicates if the device supports an INVITE request that does not contain an SDP offer. The device responds with an SDP offer in its response to the INVITE.

• true—The CCXML Device supports receiving INVITE requests that do not contain an SDP offer.

• false—The CCXML Device does not support receiving INVITE requests that do not contain an SDP offer.

options-support Support for SIP OPTIONS.Indicates if the device supports the SIP OPTIONS request. The device includes its SDP capabilities in the body of the OPTIONS response.

• true—The CCXML Device supports receiving OPTIONS requests.

• false—The CCXML Device does not support receiving OPTIONS requests.

Table 33: CCXML Device Profile Class Properties (Continued)


User’s Guide 191


recvonly-support Support for recvonly media lines.Indicates if the offer-answer user agent supports the a=recvonly media line attribute in SDP messages that it receives.

• true—The CCXML Device supports receiving the recvonly attribute in the SDP media line.

• false—The CCXML Device does not support receiving the recvonly attribute in the SDP media line.

restricts-media-source

Support for receiving media only from the User Agent (UA) to which media is being sent.Indicates if the offer-answer user agent is able to receive media from a UA other than the UA to which it is sending media.

• true—The CCXML Device supports receiving media from the UA to which media is being sent, and only from that UA.

• false—The CCXML Device supports receiving media from the UA to which media is being sent, as well as from other UAs.

sendonly-support Support for sendonly media lines.Indicates if the offer-answer user agent supports the a=sendonly media line attribute in SDP messages that it receives.

• true—The CCXML Device supports receiving the sendonly attribute in the SDP media line.

• false—The CCXML Device does not support receiving the sendonly attribute in the SDP media line.





unjoined-initial-answer-pref

Indicates the preferred method for performing an answer during an initial INVITE when no bridges have been established.

• connectionless-SDP—If the value of connectionless-sdp-type is anything other than none, a response with the specified connectionless-SDP type will be sent to the endpoint. If the value of connectionless-sdp-type is none, there is no preferred method (this parameter is treated as if the preference was none).

• reject-media—If the value of offer-answer-support is true, a response with SDP that rejects all media lines (0 port) will be sent to the endpoint. Otherwise, there is no preferred method (this parameter is treated as if the preference was none).

• nomedia-SDP—If the value of nomedia-SDP-support is true and the value of offer-answer-support is false, a response with no-media SDP will be sent to the endpoint. Otherwise, there is no preferred method (this parameter is treated as if the preference was none).

• none—No preferred method. Other device profile parameters determine the method.



User’s Guide 193


Customizing Device ProfilesVP Call Control Platform 8.0 is preprovisioned with a number of default device profiles, which reflect Genesys’ knowledge of the behavior of some commonly used SIP devices, including the GVP Media Control Platform. For details about the default device profile attributes, see Appendix D on page 304.Your deployment may require the Call Control Platform to interface with a SIP device that is not currently defined in the default device profile provisioning file. If the SIP device attributes do not match any of the preprovisioned device profiles, you must create a new device profile, or else modify an existing one, to match the actual attributes supported by the SIP device.• If the SIP request from the unknown device includes the User-Agent

header, or another header that the Call Control Platform can use to identify the device, Genesys recommends that you create a new device profile.

• If the SIP request does not include headers that the Call Control Platform can use for identification purposes, calls from the unknown device will use one of the default device profiles (Default Inbound, Default Outbound,

unjoined-initial-offer-pref

Indicates the preferred method for performing an initial INVITE without establishing bridges.

• offer-less—If the value of offer-less-invite-support is true, an INVITE without an offer will be sent to the endpoint. Otherwise, there is no preferred method (this parameter is treated as if the preference was none).

• connectionless-SDP—If the value of connectionless-sdp-type is anything other than none, an INVITE with the specified connectionless-SDP type will be sent to the endpoint. If the value of connectionless-sdp-type is none, there is no preferred method (this parameter is treated as if the preference was none).

• nomedia-SDP—If the value of both nomedia-SDP-support and options-support is true, an INVITE with no-media SDP will be sent to the endpoint. If the value of either or both nomedia-SDP-support and options-support is false, there is no preferred method (this parameter is treated as if the preference was none).

• none—No preferred method. Other device profile parameters determine the method.





Default Dialog, or Default Conference). In this case, if you wish to support the unknown device, you must modify parameters in the default device profile(s).For example, if an unknown SIP device that does not support Offer-Answer makes an inbound call to the Call Control Platform, the call will fail unless you change the offer-answer-support parameter for the Default Inbound device profile from true to false.

The following procedure describes how to modify the device profile provisioning file.

Procedure:Provisioning Device Profiles for the Call Control Platform

Purpose: To modify the ccpccxml_provision.dat file to enable the Call Control Platform to interact with non-default SIP devices.

Prerequisites

• The Call Control Platform has been installed in a directory for which you have write access permissions.

• You have identified the required attributes for the device profile(s) you want to create or modify.

Start of procedure

1. Back up the existing ccpccxml_provision.dat file, in case you later want to restore the original settings.

2. Open the <Call Control Platform Installation Directory>\config\ccpccxml_provision.dat file in a text editor.

3. Add or modify device profile entries as required for your deployment.The format for each device profile entry is the following:<entry id="<Entry ID>" type="401" name="CCXML Device Profile">

<Precedence>

<Profile Name>

<Device Profile Class Name>

<# of properties>

<Property Name 1> <Property Value 1>

Tip: To verify which device profile was used for a failed call, use the log files at debug level: Search for Select Profile, and match the incoming INVITE to the device profile selection. Then review the parameter values for that profile (see Table 74 on page 305) to identify the parameters you need to change.

User’s Guide 195


...

<Property Name m> <Property Value m>

<SIP Header Name> <Regex>

</entry>

Where:• <Entry ID> is an unsigned integer that uniquely identifies the entry.• <Precedence> is an unsigned integer that indicates the order of priority

in which the Call Control Platform will attempt to assign the device profile. The larger the value, the lower the precedence (1 is the highest). A value of 0 (zero) indicates default. Except for 0, precedence values must be unique.

• <Profile Name> is a unique alphanumeric string that identifies the device profile. Spaces are allowed. The Call Control Platform attempts to match the value of this property to CCXML hints in outbound connections, dialogs, and conferences.

• <Device Profile Class Name> is the class of device profiles to which this device profile belongs.The only class that has been defined by default is CCXML Device. For the properties and default values that have been defined for the CCXML Device class, see “CCXML Device Class” on page 189.

• <# of properties> is the number of properties that are defined in the entry.

• <Property Name x> is a non-empty alphanumeric string that must be unique within the device profile. Spaces are not allowed.

• <Property Value x> is a non-empty alphanumeric string that specifies the value of the <Property Name x> property. Spaces are not allowed.

• <SIP Header Name> is an optional parameter that, if defined, specifies the SIP header from the incoming SIP INVITE that the Call Control Platform will attempt to match, to assign the device profile for inbound connections. Spaces are not allowed.The value for the predefined CCXML Device class is User-Agent.

• <Regex> is the expression that the Call Control Platform will attempt to match in the specified SIP header from the incoming SIP INVITE. If <SIP Header Name> is empty, <Regex> is also empty.

Note: The angle brackets in the first and last lines of each device profile entry are required characters in the syntax.

4. Save the file.5. Restart the Call Control Platform.

End of procedure



User’s Guide 197

Chapter

9 Configuring the Fetching Module and Squid ProxyThis chapter describes the requirements to configure the Fetching Module and Third-Party Squid caching proxy in your Genesys Voice Platform (GVP) deployment.It contains the following sections:

Task Summary: Configuring the Fetching Module and Squid, page 197Important Fetching Module Configuration Options, page 198Configuring the Squid Caching Proxy, page 200

Task Summary: Configuring the Fetching Module and Squid

Table 34 summarizes the configuration steps and options to configure the Fetching Module and Squid Caching Proxy in your GVP deployment.


Chapter 9: Configuring the Fetching Module and Squid Proxy Important Fetching Module Configuration Options

Important Fetching Module Configuration Options

This section describes the key configuration options that you may want to customize.Configure the options in the Genesys Administrator on the Provisioning >

Environment > Applications > <Fetching Module> > Settings tab of each Fetching Module Application in your deployment. For the detailed steps to configure option settings, see Viewing or modifying GVP configuration parameters, page 78.Except for some ems options, all changes to Fetching Module options take effect on restart.

Note: If you restart the Fetching Module, you must also stop and then restart the associated Media Control Platform or Call Control Platform.

Table 34: Configuring the Fetching Module and Squid


Modify the Squid caching proxy configuration, if required for the following reasons:• To configure for a second-level

proxy.• You cannot configure the Web

Server to deliver Expires headers, and you need to change the Squid refresh-pattern rules.

• You are following the recommended practice of denying access to all ports except those that you have identified as safe, but the ports you are using for HTTP or, if applicable, HTTPS and SSL are not the ports that are configured as safe ports and SSL ports, respectively, in the default Squid configuration file.

See “Configuring the Squid Caching Proxy” on page 200.For more information about configuring Squid, which is an open-source product, see online sources.

Schedule a task to rotate the Squid Caching Proxy service logs.

See the chapter about post-installation activities in the Genesys Voice Platform 8.0 Deployment Guide.

User’s Guide 199

Chapter 9: Configuring the Fetching Module and Squid Proxy Important Fetching Module Configuration Options

The Fetching Module configuration options are in the following configuration sections:• ems (see Table 13 on page 104)—Parameters determine EMS Reporting

behavior for call detail records (CDRs) and metrics.• log (see “Configuring Logging” on page 111)—Parameters determine

behavior for Management Framework logging.• iproxy—Parameters determine the behavior of the pwproxy process (the

Fetching Module as an HTTP or HTTPS proxy).Table 35 provides information about some important Fetching Module parameters in the iproxy section. Table 35 provides parameter descriptions as well as the default parameter values that are preconfigured in the Fetching Module Application object.For information about all the available configuration options for the Fetching Module, see the Genesys Voice Platform 8.0 Configuration Options Reference.

Table 35: Selected Fetching Module Configuration Options


iproxy Section

<protocol>_proxy <protocol> is either HTTP or HTTPS.The IP address and port that the HTTP or HTTPS proxy will use.• If HTTP_proxy is disabled (empty), the pwproxy

will not use an HTTP proxy.• If HTTPS_proxy is disabled (empty), the pwproxy will not use an HTTPS proxy.

<Proxy IP address>:<port>

Default value:• HTTP proxy—127.0.0.1:3128

• HTTPS proxy—Empty

no_cache_url_substr A space-separated list of substrings that, if contained in a URL, will ensure that the page is not cached.

<Substring1>[ <Substring2>...]

Default value: cgi-bin

ssl_* Various options starting with ssl_, to configure aspects of Secure Socket Layer (SSL) functioning.For more information about configuring secure communications in GVP, see “Enabling Secure Communications” on page 93.


Chapter 9: Configuring the Fetching Module and Squid Proxy Configuring the Squid Caching Proxy

Configuring the Squid Caching ProxyIn general, the default Squid configuration file should be suitable for most installations. However, there are three reasons why you might need to modify the Squid configuration file:• You need to configure for a second-level proxy.• You cannot configure your Web Server to deliver Expires headers, and you

wish to change the Squid defaults for the expressions Squid tries to match in SIP request-URI headers to control refresh behavior.

• You need to configure non-standard “safe” ports or SSL ports for HTTP and SSL.By default, the Squid configuration file:

Identifies the following as SSL ports: port 443 563.Identifies the following as a “safe” port for HTTP: port 80.Denies requests to unknown ports (in other words, ports that are not identified as “safe”).Denies CONNECT to other than SSL ports.

The following procedure describes how to modify the Squid configuration file.

Procedure:Modifying the Squid Configuration

Purpose: To modify the configuration file of the caching proxy to enable a second-level proxy, to specify different refresh-pattern rules for matching Request-URI expressions, or to enable non-standard “safe” and SSL ports.

Perform this procedure on each Media Control Platform and Call Control Platform host in your deployment whose behavior you want to modify.

Prerequisites

• You have the required permissions to modify files in the Squid configuration directory.

Start of procedure

1. Back up the original configuration file in case you need to restore it later.2. Open the Squid configuration file (C:\squid\etc\squid.conf) in a text

editor.

Second-LevelProxy

3. To configure for a second-level proxy, add the following lines:cache_peer <parentcache.yourdomain.com> parent <port> 0 noquerydefaultacl local-servers dstdomain <yourdomain.com>

User’s Guide 201


acl all src 0.0.0.0/0.0.0.0never_direct deny local-serversnever_direct allow all

Where:• <parentcache.yourdomain.com> is the next proxy in the chain.• <port> is the port number on which the parent cache is listening.• <yourdomain.com> identifies the domains that should not go through the

parent proxy.

Refresh-PatternRules

4. To modify the Squid refresh-pattern rules, add or reorder as many lines as you require, to specify the refresh-patterns in the order in which you want Squid to consider them. Use the following format for each line:refresh_pattern [-i] regex <min> <percent> <max> [<options>]

Where:• <min> is the amount of time, in minutes, that an object without an

explicit expiry time should be considered fresh. The recommended value is 0. Any non-negative values may cause dynamic applications to be erroneously cached unless the application designer has taken the appropriate actions.

• <percent> is a percentage of the age of the object (where age is the time since last modification) that an object without an explicit expiry time will be considered fresh.

• <max> is the upper limit, in minutes, for how long objects without an explicit expiry time will be considered fresh.

• <options> are one or more of the following:• override-expire—Enforces min age even if the server sent an

Expires: header. Doing this violates the HTTP standard. Enabling this feature could make you liable for problems, which it causes.

• override-lastmod—Enforces min age even on objects that were modified recently.

• reload-into-ims—Changes client no-cache or reload to If-Modified-Since requests. Doing this violates the HTTP standard. Enabling this feature could make you liable for problems, which it causes.

• ignore-reload—Ignores a client no-cache or reload header. Doing this violates the HTTP standard. Enabling this feature could make you liable for problems, which it causes.

The default is:refresh-pattern. 0 20% 4320

For more information about how Squid uses the refresh-pattern rules to determine the freshness or staleness of an object, see “Squid Expiry Time Algorithm” on page 316.



Configure “safe”and SSL ports

5. In the ACCESS CONTROLS section:a. Add or modify access control lines as required to ensure that the

following lines match the applicable port configurations in your deployment:• acl Safe_ports port <safe port> #http

• acl Safe_ports port <safe port> #https

• acl SSL_ports port <SSL port>

b. To deny requests to unknown ports (in other words, ports that have not been identified as “safe”), verify that the following line has not been commented out or deleted:http_access deny !Safe_ports

c. To deny connections to other than SSL ports, verify that the following line has not been commented out or deleted:http_access deny CONNECT !SSL_ports

6. Save the file.

Execute theUpdate

7. Do one of the following to execute the update:• Execute the following command to force a re-read of the configuration

file:C:\squid\sbin\squid.exe -k reconfigure -n squidNT

• Restart the SquidNT service.Restarting Squid will not affect the Fetching Module. However, if a fetch is in progress, it may fail.

Note: Changes to the configuration file are not reflected in the running configuration until you execute this command.

User’s Guide 203

Chapter

10 Configuring the Reporting ServerThis chapter provides information about configuring the Reporting Server. It contains the following sections:

Task Summary: Configuring the Reporting Server, page 203Configuring Reporting, by Granularity, page 204Configuring Database Retention Policies, page 206Important Reporting Server Configuration Options, page 207Controlling Access to Reporting Services, page 210Connecting to the Genesys Administrator, page 213

Task Summary: Configuring the Reporting Server

Table 36 summarizes the configuration steps and options to set up the Reporting Server and to customize EMS Reporting behavior in your GVP deployment.

Table 36: Configuring the Reporting Server


Verify directory paths for:• Java Message Service (JMS) for

CDR reporting.• The Atomikos distributed

transactions processing engine.

If necessary, modify settings for options in the following configuration sections:• messaging. In particular, verify the path to the directory that

ActiveMQ uses for persistent queuing (activemq.dataDirectory).

• transaction.


Chapter 10: Configuring the Reporting Server Configuring Reporting, by Granularity

Configuring Reporting, by GranularityGranularity refers to the degree of aggregation in a given summary report. For example, a request for a Call Peak report at the granularity level of month will return a peak value for each month in the requested time range. A request for a Call Peak report at the granularity level of week will return a peak value for each week in the requested time range.The Reporting Server supports reporting at the following levels of granularity:• 5-minute

Configure logging. See “Configuring Logging” on page 111. Note that the Reporting Server does not support time-based log rollover.

Configure the maximum size of reports for different levels of granularity (5-minute period, hour, day, week, month).

If necessary, modify settings for the rs.query.limit.<granularity period> options in the reporting configuration section.For more information, including a summary of the default maximums, see “Configuring Reporting, by Granularity”.

Configure the maximum size of Call Detail Record (CDR) and Call Events reports.

• If necessary, modify the cdr.max-page-size option (see page 209), to configure a suitable value for your deployment, for the maximum number of CDR or metrics records per page The default is 100.

• Consider also the cdr.max-page-count option (see page 208), for the maximum number of pages per report. The default is 10.

Configure database retention policies.

If necessary, modify settings for the options in the dbmp configuration section.For more information, see “Configuring Database Retention Policies” on page 206.

Configure Reporting Server behavior in general.

See “Important Reporting Server Configuration Options” on page 207.

(Optional) Configure HTTP Basic Authorization to secure access to Reporting services.

See “Controlling Access to Reporting Services” on page 210.

Verify that the Genesys Administrator displays GVP reports requested from the Monitoring > Voice Platform navigation panel.

If necessary, configure or modify the connection between the Genesys Administrator and the Reporting Server. For more information, see “Connecting to the Genesys Administrator” on page 213.

Table 36: Configuring the Reporting Server (Continued)


User’s Guide 205

Chapter 10: Configuring the Reporting Server Configuring Reporting, by Granularity

• Hour• Day• Week• MonthIf the requested time period does not encompass an integral unit that matches the specified granularity level, then the Reporting Server expands the time to cover an integral number. For example, if the granularity is day, a request for a report from 2008/01/01 00:00 – 2008/01/01 14:00 will be expanded to 2008/01/01 00:00 – 2008/01/02 00:00.The Reporting Server normalizes From and To parameters that specify the time range in a reporting request, so that they lie on time unit boundaries that match the granularity level. For example, if the granularity is hour, then the Reporting Server normalizes the start time and end time of the report so that they point to the beginning of an hour. In this case, a start or end time request for 11:30 will be normalized to 11:00.Ensure that the values that are set for the rs.query.limit.<granularity> configuration options in the reporting section (see page 210) are appropriate for your reporting purposes and environment.Table 37 summarizes the default values for the reporting.rs.query.limit.<granularity> configuration options. Each of these options specifies the maximum number of units of a particular aggregation period that will be included in reports at that aggregation period’s level of granularity.

Table 37: Default Maximum Units, by Granularity Level

Aggregation Period Maximum Number of Units

5 minutes 288 (5-minute periods, equals 1 day)

Hour 168 (hours, equals 1 week)

Day 92 (days)

Week 53 (weeks)

Month 36 (months)


Chapter 10: Configuring the Reporting Server Configuring Database Retention Policies

Configuring Database Retention PoliciesBy default, the database maintenance process runs daily to purge data in accordance with database retention policies. The database retention policies are defined in the following options in the dbmp configuration section:• On the Reporting Server, the rs.db.retention.*.default options (see

page 209)—These set the default retention periods for the GVP deployment overall.

• On the IVR Profile, equivalent rs.db.retention.* options—These override the default retention periods, for data relating to the specificVoiceXML or CCXML application.

Table 38 summarizes the default Reporting Server database retention periods for data at the varying levels of granularity (aggregation periods).Before you modify the default retention periods, consider your reporting requirements and the reporting results you expect. Ensure that your default database retention period settings are consistent with settings for the reporting.rs.query.limit.<granularity> configuration options, so that data that you expect to include in reports at various granularity levels is not purged prematurely from the database.

Table 38: Default Database Retention Periods

Type of Data Granularity Option Name in dbmp Section Minimum Valid Value (integer)

Default Value, in days

CDRs N/A rs.db.retention.cdr.default > 0 30

Call log events (upstream logs)

N/A rs.db.retention.events.default > 0 7

Operational data

5-minute rs.db.retention.operations.5min.default > 0 1

Daily rs.db.retention.operations.daily.default > 30 90

Hourly rs.db.retention.operations.hourly.default > 0 7

Monthly rs.db.retention.operations.monthly.default

> 30 1095 (36 months)

Weekly rs.db.retention.operations.weekly.default

> 30 364 (52 weeks)

User’s Guide 207

Chapter 10: Configuring the Reporting Server Important Reporting Server Configuration Options

Important Reporting Server Configuration Options


Environment > Applications > <Reporting Server> > Settings tab. For the detailed steps to configure option settings, see Viewing or modifying GVP configuration parameters, page 78.The configurable Reporting Server parameters are in the following configuration sections:• cdr—Parameters determine behavior for processing and reporting on call

detail records (CDRs).• dbmp—Parameters determine database retention policies.• log (see “Configuring Logging” on page 111)—Parameters determine

logging behavior.• messaging—Parameters specify the paths for the ActiveMQ JMS broker

that receives Reporting Server messages.• persistence—Parameters configure behavior for Hibernate interactions

with the database.• reporting—Parameters determine the number of records that will be

considered for different levels of granularity.• schedule—Parameters provide the cron expressions for scheduling

periodic tasks.• transaction—Parameters provide the directory paths for the Atomikos

distributed transactions processing engine.

VAR summary statistics (Call Summary and IVR Action statistics)

5-minute rs.db.retention.var.5min.default > 0 1

Daily rs.db.retention.var.daily.default > 30 90

Hourly rs.db.retention.var.hourly.default > 0 7

Monthly rs.db.retention.var.monthly.default > 30 1095 (36 months)

Weekly rs.db.retention.var.weekly.default > 30 364 (52 weeks)

Table 38: Default Database Retention Periods (Continued)

Type of Data Granularity Option Name in dbmp Section Minimum Valid Value (integer)

Default Value, in days



Table 39 provides information about important Reporting Server parameters that are not described in Chapter 4 on page 85. Table 39 provides parameter descriptions as well as the default parameter values that are preconfigured in the Reporting Server Application object.Except for changes in the dbmp and log sections, all changes take effect on restart.For information about all the available configuration options for the Reporting Server, see the Genesys Voice Platform 8.0 Configuration Options Reference.

Table 39: Selected Reporting Server Configuration Options


cdr Section

call-timeout The amount of time, in minutes, until a call is considered timed out from the perspective of VAR and CDR reporting.The Reporting Server may receive no CDR call-termination update because:• The call was dropped from the platform (for

example, because a component shut down unexpectedly).

• The Reporting Server is simply not receiving updates from the component (for example, because the network connection is down). The component queues data that it cannot send to the Reporting Server, so the Reporting Server may eventually receive a CDR update for a call that was previously assumed to be timed out. In these cases, the Reporting Server will appropriately update the CDR.

The interval at which the timeout process runs is configurable (see schedule.quartz.rs.calltimeout on page 210). The timeout process uses the value of the call-timeout parameter to identify calls that have timed out since the process last ran.

An integer in the range of 1-1440.Default value: 180 (3 hours)

max-page-count The maximum number of pages that will be returned in any given CDR or Call Events report request.


User’s Guide 209


max-page-size The maximum number of records that will be returned in a single page in any given CDR or Call Events report request.This limit prevents users from overloading the system by requesting unreasonably large numbers of CDRs or metrics in a report.

An integer in the range of 1-10000.Default value: 100

dbmp Section

rs.db.retention.<type of data>.default

The number of days for which <type of data> will be retained in the Reporting database, where <type of data> is:• cdr—CDR data• events—call log events (upstream logs) data• operations.5min—5-minute operational data• operations.daily—daily operational data• operations.hourly—hourly operational data• operations.monthly—monthly operational

data• operations.weekly—weekly operational data• var.5min—5-minute Call Summary and IVR

Profile Summary data• var.daily—daily Call Summary and IVR

Profile Summary data• var.hourly—hourly Call Summary and IVR

Profile Summary data• var.monthly—monthly Call Summary and

IVR Profile Summary data• var.weekly—weekly Call Summary and IVR

Profile Summary dataNote: You can specify non-default dbmp.rs.db.retention.<type of data> options for individual IVR Profiles. If configured, the IVR Profile rs.db.retention.<type of data> value overrides the value configured in the Reporting Server dbmp.rs.db.retention.<type of data>.default option.

See Table 38 on page 206.

Table 39: Selected Reporting Server Configuration Options (Continued)



Chapter 10: Configuring the Reporting Server Controlling Access to Reporting Services

Controlling Access to Reporting ServicesGVP 8.0 leverages the HTTP Basic Authorization features of Apache Tomcat to authenticate users and control access to Reporting Web Services.

reporting Section

rs.query.limit.<time period>

The maximum number of <granularity> periods that are included in any report with a granularity of <granularity>, where <granularity> is:• 5min—5-minute periods• day

• hour

• month

• week

If a reporting request at a particular granularity level specifies a time range that is greater than the configured maximum, the request is truncated to cover the maximum allowed time period, starting from the From time specified in the request.

See Table 38 on page 206.Example:If rs.query.limit.5min=288 (1 day), a request for a report at 5-minute granularity for the time period 2008/01/01 00:00 – 2008/01/02 12:00 will be truncated to 2008/01/01 00:00 – 2008/01/02 00:00.

schedule Section

quartz.rs.calltimeout The cron schedule for Quartz to execute the Call Timeout Process, which is responsible for timing out Resource Manager, Media Control Platform, Call Control Platform, and VAR CDRs, so that they do not get stuck as open calls in the database.By default, the process runs every 50 minutes.A configurable option specifies the timeout interval that determines when a call is considered timed out (see the cdr.call-timeout option on page 208).

See http://www.opensymphony.com/quartz/api/org/quartz/CronExpression.html and http://quartz.sourceforge.net/javadoc/org/quartz/CronTrigger.html.Default value: 0 50 * * * ?

quartz.rs.dbMaintenancePeriod

The cron schedule for Quartz to purge data from the database, in accordance with data retention policies.By default, the process runs at 1 a.m. every day.

See http://www.opensymphony.com/quartz/api/org/quartz/CronExpression.html.

Default value: 0 0 1 * * ?

Table 39: Selected Reporting Server Configuration Options (Continued)


User’s Guide 211


GVP 8.0 does not support selective, role-based access for different categories of Reporting services.The following procedure describes how to configure Tomcat and Management Framework so that the Genesys Administrator will silently provide authentication information to the Reporting Server when it invokes Reporting Web Services.

Procedure:Enabling HTTP Basic Authorization for Reporting

Purpose: To enable HTTP Basic Authorization for Reporting, by modifying the configurations of Tomcat and the Management Framework user interface (UI).

Start of procedure

1. Shut down the Apache Tomcat process.2. Define the user name and password that the Genesys Administrator will be

required to provide for authentication, in order to access Reporting services.a. Open the <Tomcat Home Directory>\conf\users.xml file.b. Edit the file, to specify the required role, user name, and password.

The lines in bold text in the following example show the modifications for the Reporting Web Services client role, EMS Reporting.(Contrary to type conventions in the remainder of this guide, italic text indicates placeholders for user-specified values, user-name and password. The angle brackets are a required part of the XML syntax.)

<?xml version='1.0' encoding='utf-8'?>

<tomcat-users>



<role rolename="EMS Reporting"/>



<user username="user-name" password="password" roles="EMS Reporting"/></tomcat-users>

3. Update the web service configuration.a. Open the web application configuration file:

<Tomcat Home Directory>\webapps\ems-rs\WEB-INF\web.xml

b. Edit the file, to specify that authentication is required and that the EMS Reporting role has access to Reporting services.The lines in bold text in the following example show the modifications to grant access to EMS Reporting users.



<?xml version="1.0" encoding="UTF-8"?>

<web-app xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance"

xmlns="http://java.sun.com/xml/ns/javaee"

xmlns:web="http://java.sun.com/xml/ns/javaee/web-app_2_5.xsd"

xsi:schemaLocation="http://java.sun.com/xml/ns/javaee

http://java.sun.com/xml/ns/javaee/web-app_2_5.xsd"

version="2.5">

<servlet>

<servlet-name>RestServlet</servlet-name>

<servlet-class>

com.noelios.restlet.ext.servlet.ServerServlet

</servlet-class>

<load-on-startup>1</load-on-startup>

</servlet>



<context-param>

<param-name>org.restlet.application</param-name>

<param-value>

com.gvp.rpt.service.ReportingServiceContext

</param-value>

</context-param>

<servlet-mapping>

<servlet-name>RestServlet</servlet-name>

<url-pattern>/*</url-pattern>

</servlet-mapping>



<security-constraint>

<web-resource-collection>

<web-resource-name>Reporting Services</web-resource-name>

<url-pattern>/*</url-pattern>

</web-resource-collection>

<auth-constraint>

<role-name>EMS Reporting</role-name>

</auth-constraint>

</security-constraint>



<login-config>

<auth-method>BASIC</auth-method>

<realm-name>EMS Reporting</realm-name>

</login-config>



<security-role>

<description>Reporting Services role</description>

<role-name>EMS Reporting</role-name>

User’s Guide 213

Chapter 10: Configuring the Reporting Server Connecting to the Genesys Administrator

</security-role>

</web-app>

4. Restart Apache Tomcat.Apache Tomcat will now authenticate all web service requests, including those from the Management Framework Reporting UI (the Genesys Administrator).

5. Update the EMS Reporting user name and password in the Management Framework Reporting UI configuration. The relevant options are in the rptui configuration section of the default Application object.a. In the Genesys Administrator, go to the Provisioning > Environment >

Applications > default > Settings tab.b. From the display filter drop-down list, select GVP Reporting.

The Settings tab displays the configuration options for the GVP Reporting UI (rptui section).

c. Modify the rptui.rs.username and rptui.rs.password options, to specify the user name and password you configured for Tomcat authorization (see Step 2 on page 211).

d. Click Save or Apply.6. Restart the web server, or else log out of the Genesys Administrator and

then log back in.7. Verify that GVP reports display in the Genesys Administrator when you

request them from the Monitoring > Voice Platform navigation panel..

End of procedure

Connecting to the Genesys AdministratorThe following procedure describes the steps to configure the connection between the Genesys Administrator GVP Reporting UI (GVP Reports) and the Reporting Server.

Procedure:Configuring the connection between the Reporting Server and the Genesys Administrator

Purpose: To configure Management Framework to enable GVP reports to be displayed in the Genesys Administrator.



Prerequisites


Start of procedure

1. In the Genesys Administrator, go to the Provisioning > Environment > Applications > default > Settings tab.

2. From the display filter drop-down list, select GVP Reporting.The Settings tab displays the configuration options for the GVP Reports UI (rptui section).

3. Verify that the configuration option values are suitable for your deployment.• If you have configured Tomcat for HTTP Basic Authorization on the

Reporting Server, ensure that you specify the user name and password (rs.username and rs.password) to enable GVP Reports to provide authentication information to Tomcat. For more information, see Enabling HTTP Basic Authorization for Reporting, page 211.

• To enable HTTPS communication between the Reporting Server and the Genesys Administrator, set the enablehttps option to true.

For more information about the rptui configuration options, see Table 40.4. Click Save.5. On the Provisioning > Environment > Applications > default > General

tab, add a connection to the Reporting Server. For more information, see the procedure about creating a connection to a server in the Genesys Voice Platform 8.0 Deployment Guide.

6. Restart the web server, or else log out of the Genesys Administrator and then log back in.

End of procedure

Table 40 provides information about the options in the rptui configuration section of the default Application object. Table 40 provides parameter descriptions as well as the default parameter values that are preconfigured in the default Application object.The default Application object is created automatically and is always available when you start the Genesys Administrator.

User’s Guide 215


Table 40: Reporting UI Configuration Options—default Application


dsthours The daylight savings time difference, in hours and minutes, to apply to timestamps to adjust to local time.When specifying the value, use leading zeros if necessary.

<HH>:<mm>

where:• <HH> indicates hours.• <mm> indicates minutes.Default value: 01:00

enablehttps Specifies whether GVP Reports will use HTTP or HTTPS to access Reporting Web Services.

• true—HTTPS will be used.

• false—HTTP will be used.

Default value: false

httptimeout The timeout, in seconds, for communications between GVP Reports and the Reporting Server.If your deployment experiences frequent timeouts, increase this value.

Any positive integer.Default value: 30 (seconds)

localtimeformat Specifies whether GVP Reports will display date and time values in local time, rather than in Greenwich Mean Time (GMT), which is the default format that Reporting Server returns.To display local time in reports, set this option to true (1) and specify a timezone offset (see tzoffset).

• true (1)—Date and time values will display in local time.

• false—Date and time values will display in GMT.

Default value: true

rs.httpport The Tomcat port where Reporting Web Services are available. GVP Reports uses this port to retrieve data from the Reporting Server.The value of this option overrides the server port configured in the Reporting Server Application. If this option is not specified in the default Application, the port configured in the Reporting Server Application is used (default is 8080).


rs.password The password to be provided to the Reporting Server, together with the configured user name (rs.username), if the Reporting Server web server has been configured to authenticate user credentials.For more information, see Enabling HTTP Basic Authorization for Reporting, page 211.




rs.username The user name to be provided to the Reporting Server, together with the configured password (rs.password), if the Reporting Server web server has been configured to authenticate user credentials.For more information, see Enabling HTTP Basic Authorization for Reporting, page 211.


tzoffset The time offset, in hours and minutes, that will be applied to convert GMT to local time (see localtimeformat), in the timezone where GVP reports will be accessed.Dates and times in all GVP reports will be converted.When specifying the value, use leading zeros if necessary.

<s><HH>:<mm>

where:• <s> is either + (plus) or -

(minus), to indicate whether the time should be added or subtracted.

• <HH> indicates hours.• <mm> indicates minutes.Default value: -08:00

Table 40: Reporting UI Configuration Options—default Application (Continued)


User’s Guide 217

Part

3 Monitoring GVPThis part of the guide describes the available real-time and historical reports in the Genesys Administrator.This information appears in the following chapters:• Chapter 11, “Reporting Overview,” on page 219• Chapter 12, “Real-Time Reports,” on page 227• Chapter 13, “Historical Reports,” on page 233• Chapter 14, “Voice Application Reports,” on page 243


User’s Guide 219

Chapter

11 Reporting OverviewThis chapter describes how to use the Genesys Administrator to create real-time and historical reports.It contains the following sections:

Genesys Administrator, page 219Running a Report, page 219Report Filters, page 224

Genesys AdministratorThe Genesys Administrator is the tool that you use to monitor your call-center activity; it enables you to analyze call volumes, trends, and the effectiveness of your voice and call-control applications.For more information on how to use Genesys Administrator, see the Framework 8.0 Genesys Administrator Help file.

Running a ReportThe following procedure explains how to run a report using Genesys Administrator.

Procedure:Running a Report

Purpose: To run a report by using the Genesys Administrator.


Chapter 11: Reporting Overview Running a Report

Prerequisites

• The valid URL for Genesys Administrator—for example, http://<Genesys Administrator host>/wcm/.

• The username and password with the correct permissions for creating reports.

• The name of the Genesys Administrator application—for example, default.

Note: The default application object is automatically created, and is seen when Genesys Administrator is invoked.

• The host name and port of the Genesys Configuration Server.

Start of procedure:

1. In the web browser’s address bar, enter http://<Genesys Administrator host>/wcm/.

The login to Genesys Administrator dialog box appears.2. Enter the following parameters:

• User Name• Password• Application• Host Name• Port

3. Click Login.The Genesys Administrator screen appears (see Figure 9) with the Monitoring tab active.

User’s Guide 221


Figure 9: Genesys Administrator

4. From the Navigation panel, select Voice Platform.The reporting categories become visible (see Figure 10).

Figure 10: Genesys Administrator Voice Platform Reporting Categories



5. Select the required reporting category.The list of available reports is visible in the right-hand panel of Genesys Administrator (for example, see Figure 11, the Historical Report List).

Figure 11: Historical Report List

6. Select the required report from the list of available reports.The Configure Filters screen appears with the possible filter criteria (see Figure 12). A different set of filters appear for each report type.

Figure 12: Configure Filters Screen for the Historical Call Browser Report

User’s Guide 223


7. Select the required filters. Table 41, “Reporting Filters,” on page 224 describes each filter.• Table 42 on page 228 lists the real-time reporting supported filters.• Table 45 on page 234 lists the historical reporting supported filters, and

the valid filter combinations for each report.• Table 50 on page 244 lists the VAR reporting supported filters, and the

valid filter combinations for each report.8. Click Generate Report.

The Report Results screen appears (see Figure 13).

Figure 13: Example of Historical Call Browser Report Results

9. If the report has drill down functionality, select the required record to view the details of that record (see Figure 14).

Figure 14: Example of Historical Call Browser Drill-Down Results

End of procedure


Chapter 11: Reporting Overview Report Filters

Report FiltersTable 41 describes the filter criteria that you can use to retrieve call detail records, IVR action data, or summary data for the Real-Time, Historical, and Voice Application reports.

Table 41: Reporting Filters

Filter Name Description

Filter by Date-Time

Filters the data by start date, end date, start time, and end time. The results will display calls that started on or after the start time and ended before the end time.

Filter by IVR Profile

Filters the data by IVR Profile. You can choose more than one IVR Profile for some of the reports. For more information, see the individual reports.For more information on IVR Profiles, see Chapter 6 on page 139.

Filter by Call End State

Filters the data by Call End States. The possible Call End States are:• Application End—The voice application hung up.• System Error—The call did not end properly.• Unknown—The MCP did not log an end state.• User End—The caller hung up.

Filter by Call Result

Filters the data by Call Results. The possible Call Results are:• Success—The call was processed successfully.• Failed—A failure occurred that prevented the call

from being processed properly.• Rejected—The call was rejected by MCP.• Unknown—Some unknown reason caused the call to end

abruptly.

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Filter by ID’s Filters the data by the call ID. The possible IDs are:• Session ID—The GVP Component specific ID that is

generated by the component to identify the call leg.• GVP GUID—The globally unique ID that identifies a

complete interaction with GVP. This ID is generated by the Resource Manager, and is passed to all the resources that provide service for the call.

• Genesys ID—The Genesys CallUUID that is generated by T-Server or SIP Server.

For more information on these IDs, see Chapter 1, “Introduction,” on page 21.

Filter by Component

Filters the data by the component. A component is a provisioned Resource Manager, MCP, or CCP application. You can choose more than one component for some of the reports. For more information, see the individual reports.Note: All selected components must be of the same type—RM, CCP, or MCP.

Filter by Granularity Level

Presents the data at various levels of aggregation:• Five Minutes

• Hour

• Day

• Week

• Month

Filter by Call Type

Filters the data by call type. The possible call types are:• Inbound—Applicable for MCP and RM components.• Outbound—Applicable for MCP and RM components.• Bridged—Applicable for MCP components only.• NewCall—Applicable for CCP components only.• Create-CCXML—Applicable for CCP components only.• External—Applicable for CCP components only.• Unknown—Applicable for RM components only.

Table 41: Reporting Filters (Continued)




The data on the reports that use the granularity filter are stored in the database for the length of time that is given for the dbmp.rs.db.retention.cdr.default parameter. Granularity works with the data reporting limits that are configured in the Reporting Server. These limits are the maximum amount of data that the Reporting Server returns based on the which granularity level is selected. The Report Server options are:• rs.query.limit.5mins

• rs.query.limit.hour

• rs.query.limit.day

• rs.query.limit.week

• rs.query.limit.month

For more information , see “Configuring Reporting, by Granularity” on page 204.

Filter by Call Length

Filters the data by the length of time, in milliseconds, of the call. Minimum and maximum durations can be specified. If only a minimum duration is specified, calls that exceeded this duration are displayed. If only a maximum duration is specified, calls that lasted for less than or equal to this duration are displayed.

Filter byRemote URI

Filters the data by the full URI of the remote party that is involved in the session.Note: Accepts the * wildcard.

Filter by Local URI

Filters the data by the URI of the local service.Note: Accepts the * wildcard.

Table 41: Reporting Filters (Continued)


User’s Guide 227

Chapter

12 Real-Time ReportsThis chapter describes the available reports that display real-time data. It contains the following sections:

Overview, page 227Active Call List, page 228

OverviewThe real-time reports display statistics of the current call that is in progress. However, real-time data updates are not instantaneous, because there may be a slight delay while the Media Control Platform (MCP), the Call Control Platform (CCP), or Resource Manager (RM) sends data to the Reporting Server.Call detail records (CDR) and call events are delivered in batches to the Reporting Server. By default the batch size is 500 CDRs or ten seconds. This means that a message will be sent either when 500 CDR updates are queued, or ten seconds has expired, whichever occurs first. You can reconfigure the system to be more real-time by changing the batch size—for example, change it to 1. This means that a CDR update or the call event will be delivered to the Reporting Server as soon as it is raised by the component. There are performance implications to changing the batch size (for example, more transaction overhead on the server side), but the result is more real-time. The following procedure describes how to run a real-time report.

Procedure:Running a Real-Time Report

Purpose: To run a real-time report by using the Genesys Administrator.


Chapter 12: Real-Time Reports Active Call List

Start of procedure:

• Follow the instructions to create a report (see Running a Report, page 219).

• For Step 7 in those instructions, see Table 42 for the available real-time report filters.

End of procedure

Active Call ListThe Active Call List report (see Figure 15) displays the list of calls that currently are being processed by GVP. It includes also any call that the Reporting Server has not marked as timed out.A call is considered as timed out if the call processing component has unexpectedly shut down, of if the connection between the call processing component and the Reporting Server is broken. In either case, the Reporting Server has not received the update indicating that the call ended. If the connection is down, the Reporting Server will eventually receive the update; however, if the processing component unexpectedly shut down, the call will stay as timed out.The Reporting Server processes timed out calls once hourly (by default), and marks calls as timed out when they have been in progress for more than a configured period of time (by default 3 hours).

Table 42: Real-Time Report Filters

Filter Available Real-Time Reports

Active Call List


Filter by Component

User’s Guide 229


Notes: If you do not select an IVR Profile, all IVR Profile data is displayed. If you do not select a Component, only RM call detail records will display. You must select the MCP component to view MCP call detail records.Data is returned if no filter is selected.You can select multiple IVR Profiles and multiple Components.

For more information on the details of completed calls, see “Historical Call Browser” on page 238.

Figure 15: Active Call List Report

Table 43 describes the fields for the summary level for the Active Call List report.

Table 43: Active Call List Report Summary Fields

Field Description

GVPID The globally unique ID that identifies a complete interaction with GVP. This ID is generated by the Resource Manager, and is passed to all the resources that provide service for the call. For more information on the GVP Session ID, see “Session Identifiers” on page 72.

SessionID The GVP Component ID that is generated by the component to identify the call leg. For more information on the GVP Component ID, see “Session Identifiers” on page 72.



The drill-down report breaks down the detail recorded of the selected call according to component type (see Figure 16).

Figure 16: Active Call List Drill Down Report

Table 44 describes the fields for the Active Call List drill down report.

GenesysID The Genesys CallUUID that is generated by T-Server or SIP Server. For more information on the Genesys CallUUID, see “Session Identifiers” on page 72.

IVRProfile The name of the IVR Profile that is selected.

Start DateTime The start date and start time of the call.

Component The name of the Resource Manager that is selected.

Call Type The type of the call. Valid call types are the following:• INBOUND• OUTBOUND• BRIDGED• UNKNOWN• NEW CALL• CREATE-CCXML• EXTERNAL

RemoteURI The remote Uniform Resource Identifier.

LocalURI The local Uniform Resource Identifier.

Table 43: Active Call List Report Summary Fields (Continued)

Field Description

User’s Guide 231


Table 44: Active Call List Drill Down Report Fields

Fields Description

Component Type The type of component. The possible components are:• RM (Resource Manager)• MCP (Media Control Platform)• CCP (Call Control Platform)




IVR Profile The name of the IVR Profile as seen in Genesys Administrator or Configuration Manager.

Component The name of the Component application as seen in Genesys Administrator or Configuration Manager.


End DateTime The end date and end time of the call if the call has completed.



Call Status The current state of the call. Valid call states are the following:• IN-PROGRESS• TIMED OUT• COMPLETED

Call Type The type of call. Valid call types are the following:• INBOUND• OUTBOUND• BRIDGED• UNKNOWN• NEW CALL• CREATE-CCXML• EXTERNAL

Table 44: Active Call List Drill Down Report Fields (Continued)

Fields Description

User’s Guide 233

Chapter

13 Historical ReportsThis chapter describes the available historical reports. It contains the following sections:

Overview, page 233Historical Call Summary, page 235Historical Peaks, page 236Historical Call Browser, page 238

OverviewThe historical reports display call detail records, call arrival and summary information over a selected period of time, based on a set of selected criteria. The Historical Call Summary and Historical Peaks reports display the data in both a pictorial graph and a table. The graph provide the following navigation features:

To zoom in, drag from left to right on a selected area.To pan around the graph, right-click and drag the graph.To restore to normal view (100%), double-click the graph.To turn on or off the visibility of the individual series, click the eye icon in the chart legend.

The following procedure describes how to run a historical report.

Procedure:Running a Historical Report

Purpose: To run a historical report by using the Genesys Administrator.


Chapter 13: Historical Reports Overview

Start of procedure:

• Follow the instructions to run a report (see Running a Report, page 219).

• For Step 7 in those instructions, see Table 45 for the available historical filters and the valid filter combinations.

End of procedure

The data on the reports that use the granularity filter are stored in the database for the length of time that is given for the dbmp.rs.db.retention.cdr.default parameter. The retention policy can be overridden on the IVR Application by specifying the corresponding configuration in the IVR Profile (minus the .default suffix):• rs.db.retention.operations.5min.default

• rs.db.retention.operations.hourly.default

• rs.db.retention.operations.daily.default

Table 45: Historical Filters and Valid Filter Combinations

Filter Available Historical Reports

Call Summary Peaks Call Browser

Filter by Date-Time


Filter by ID’s

Filter by Component


Filter by Call Type

Filter by Call Length

Filter by Remote URI

Filter by Local URI

Filter Combinations • Date-Time, IVR Profile, Granularity

• Date-Time, Component, Granularity

• Date-Time, IVR Profile, Granularity

• Date-Time, Component, Granularity

• NoneNote: If you filter by the Session ID, you also need to provide a Component.

User’s Guide 235

Chapter 13: Historical Reports Historical Call Summary

• rs.db.retention.operations.weekly.default

• rs.db.retention.operations.monthly.default

For more information on this and other data retention parameters, see “Configuring Database Retention Policies” on page 206.

Historical Call SummaryThe Historical Call Summary report (see Figure 17) lists a summary of call arrival data that is submitted by each component for a given period of time, IVR Profile, and/or Component.

Note: You can select multiple IVR Profiles or multiple Components.

Figure 17: Historical Call Summary Report

Table 46 describes the fields for the Historical Call Summary report.

Table 46: Historical Call Summary Report Fields

Field Description

Call Date The start date and start time for the calls.


Chapter 13: Historical Reports Historical Peaks

The Daily Call Summary Details report (see Figure 18) displays the call distribution details of the selected day in five-minute intervals.

Figure 18: Historical Call Summary Details Report

The Historical Call Summary report provides a summary row at the bottom of the grid that displays the grand totals for each column.

Historical PeaksThe Historical Peaks report (see Figure 19) provides the peak volume of calls that is observed during a given period of time.

Note: You can select multiple Components; however, you can select only one IVR Profile.

Call Date End The end date and end time for the calls.

Call Volume (Inbound/Outbound/Unknown/Bridged)

The number of calls according to call type for the respective call date.

Table 46: Historical Call Summary Report Fields (Continued)

Field Description

User’s Guide 237

Chapter 13: Historical Reports Historical Peaks

The Peaks Volume, which is shown on the graph, counts the peak number of calls that is observed during the specified time range, according to the selected granularity level.

Figure 19: Historical Peaks Report

Table 47 describes the fields for the Historical Peaks report.

Table 47: Historical Peaks Reports Fields

Field Description

Peaks Date The date and time of the call for the given granularity period.

Peaks Volume The number of calls for the given time period.


Chapter 13: Historical Reports Historical Call Browser

Historical Call BrowserThe Historical Call Browser report (see Figure 20) displays a list of completed calls. It provides the ability to search for and browse call detail records. These records represent calls that either completed successfully or eventually were timed out by the Reporting Server.A call is considered as timed out if the call processing component has unexpectedly shut down, of if the connection between the call processing component and the Reporting Server is broken. In either case, the Reporting Server has not received the update indicating that the call ended. If the connection is down, the Reporting Server will eventually receive the update; however, if the processing component unexpectedly shut down, the call will stay as timed out.The Reporting Server processes timed out calls once hourly (by default), and marks calls as timed out when they have been in progress for more than a configured period of time (by default 3 hours).

Notes: You can select multiple IVR Profiles and multiple Components.Data is returned if no filter is selected.

Figure 20: Historical Call Browser Report

User’s Guide 239


Table 48 describes the fields for the Historical Call Browser report.

Table 48: Historical Call Browser Report Fields

Field Description






End DateTime The end date and end time of the call.

Duration The length of the time of the call in milliseconds.



Component The name of the application (RM, CCP, MCP) that is selected.


Call Status The state of the call. Valid call states are the following:• COMPLETED• TIMED OUT



The drill-down report breaks down the selected record according to component type. It displays the call detail records for all components that were involved in handling the call (see Figure 21). There can be multiple call detail records for each component if there was more than one leg in the call.

Figure 21: Historical Call Browser Drill-Down Report

Table 49 describes the fields for the Historical Call Browser Drill-Down report.

Table 49: Historical Call Browser Report Drill-Down

Field Description

Component Type The type of component. Valid types are:• CCP (Call Control Platform).• MCP (Media Control Platform).• RM (Resource Manager).


Genesys ID The Genesys CallUUID that is generated by T-Server or SIP Server. For more information on the Genesys CallUUID, see “Session Identifiers” on page 72.

Session ID The GVP Component ID that is generated by the component to identify the call leg. For more information on the GVP Component ID, see “Session Identifiers” on page 72.

Application ID The name of the IVR Profile as seen in Genesys Administrator or Configuration Manager.

Component ID The name of the Component application as seen in Genesys Administrator or Configuration Manager.


End DateTime The end date and start time of the call.

User’s Guide 241


Call Status The state of the call. Valid call states are the following:• COMPLETED• TIMED OUT


Table 49: Historical Call Browser Report Drill-Down (Continued)

Field Description



User’s Guide 243

Chapter

14 Voice Application ReportsThis chapter describes the Voice Application Reports.It contains the following sections:

Overview, page 243VAR Call Browser, page 244Call Completion Summary, page 247IVR Action Usage, page 248Last IVR Action Used, page 249

OverviewThe Voice Application Reports display the usability data for applications that have been divided into logical transactions.The following procedure describes how to run a VAR report.

Procedure:Running a Voice Application Report

Purpose: To run a Voice Application Report by using the Genesys Administrator.

Start of procedure:

• Follow the instructions to create a report (see Running a Report, page 219).

• For Step 7 in those instructions, see Table 50 for the available Voice Application Report filters and the valid filter combinations.

End of procedure


Chapter 14: Voice Application Reports VAR Call Browser

Table 50: Voice Application Filters and Valid Filter Combinations

VAR Call BrowserThe VAR Call Browser report (see Figure 22) provides the ability to search and browse call data that relates to VAR reporting. It displays a list of all calls that occurred for a selected period of time.

Notes: The VAR Call Browser report displays MCP data only.Data is returned if no filter is selected.Any call detail record that contains custom variables will have a paper clip icon in the second column.

Filter Available Voice Application Reports

Call Browser Call Completion Summary

IVR Action Usage

Last IVR Action Used

Filter by Date-Time


Filter by Call End State

Filter by Call Result

Filter by ID’s

Filter by Component


Filter Combinations none • DateTime, IVRProfile,Granularity

Note: You can select only one IVRProfile at a time.

• DateTime, IVRProfile,Granularity

• DateTime, IVRProfile,Granularity

Note: You can select only one IVRProfile at a time.

User’s Guide 245


Figure 22: VAR Call Browser Report

Table 51 describes the fields for the VAR Call Browser report.

Table 51: VAR Call Browser Report Fields

Field Description







Component The name of the Resource Manager application that is selected.


End DateTime The end date and end time of the call.

Duration The length of the time of the call in milliseconds.



State The end state of the call. Valid states are:• APPLICATION END—The application hung up.• SYSTEM ERROR—The call did not end properly.• UNKNOWN—The MCP did not log an end state.• USER END—The caller hung up.

Result The end result of the call, as reported by the application. Valid results are:• SUCCESS—The call was processed successfully.• FAILED—A failure occurred that prevented the call

from being processed properly—for example, a database error or network error.

• REJECTED—The MCP rejected the call—for example, the VoiceXML application could not be fetched from the application server.

• UNKNOWN—Some unknown reason caused the call to end abruptly.

Reason A string identifier, of no more than 256 characters in length, that explains the results.

Table 51: VAR Call Browser Report Fields (Continued)

Field Description

User’s Guide 247

Chapter 14: Voice Application Reports Call Completion Summary

Call Completion SummaryThe Call Completion Summary report (see Figure 23) displays the total number of calls and the percentage of calls, grouped by Call End Action data. The Call End Action displays the IVR Results and the IVR Result Reasons.

Note: The Call Completion Summary report displays MCP data only.

Figure 23: Call Completion Summary Report

Notes A string identifier, of no more than 256 characters in length, that has additional information that relates to the call.

Custom Variables The name and value of any custom variable that associates name/value pairs to the call.

Table 51: VAR Call Browser Report Fields (Continued)

Field Description


Chapter 14: Voice Application Reports IVR Action Usage

Table 52 describes the fields for the Call Completion Summary report.

IVR Action UsageThe IVR Action Usage report (see Figure 24) displays statistics on individual IVR Actions that are used for a given IVR Profile within a given time period.

Table 52: Call Completion Summary Report Fields

Field Description

Call Completion Time

The date and time (in yyyy-mm-dd hh:mm:ss format) when the call finished.

Call End State The end state of the call. Valid states are:• APPLICATION END—The application hung up.• SYSTEM ERROR—The call did not end properly.• UNKNOWN—The MCP did not log an end state.• USER END—The caller hung up.

Call End Result The end result of the call, as reported by the application. Valid results are:• SUCCESS—The call was processed successfully.• FAILED—A failure occurred that prevented the call




Reason A string identifier, of no more than 256 characters in length, that explains the result.

Total Calls The total number of calls that ended for the time duration (granularity) that is selected.

% of Calls The percentage of calls for each level of granularity with respect to the next higher grouping.

Avg. Call Length The average length, in milliseconds, of the time of the call.

User’s Guide 249

Chapter 14: Voice Application Reports Last IVR Action Used

Note: The IVR Action Usage report displays MCP data only.

Figure 24: IVR Action Usage Report

Table 53 describes the fields of the IVR Action Usage report.

Last IVR Action UsedThe Last IVR Action Used report (see Figure 25) displays the details of the last IVR Actions that were used during the end of a call.

Table 53: IVR Action Usage Report Fields

Field Description

IVRProfile The name of the IVR Profile for which these actions occurred.

IVR Action The name of the IVR Action.

Usage Count The number of times that the IVR Action was used.

% actions that were successful

The percentage of actions that were successful.

Calls that used this action

The number of calls that used this IVR Action at least once.

% age of Calls that used this action

The percentage of total calls that used this IVR Action at least once.



Note: The Last IVR Action Used report displays MCP data only.

Figure 25: Last IVR Action Used Report

Table 54 describes the fields for the Last IVR Action Used report.

Table 54: Last IVR Action Used Report Fields

Field Description

IVR Action The name of the IVR Action.

Last Used in Calls

The total number of calls in which the last IVR Action was used.

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% of Calls The percentage of the total number of calls in which this IVR Action was used.

Call Result The percentage of calls that used the Last IVR Action, with the given End Action. Valid Call Results are:• SUCCESS—The call was processed successfully.• FAILED—A failure occurred that prevented the call




Valid End Actions:• APPLICATION END—The application hung up.• SYSTEM ERROR—The call did not end properly.• UNKNOWN—The MCP did not log an end state.• USER END—The caller hung up.

Table 54: Last IVR Action Used Report Fields (Continued)

Field Description



User’s Guide 253

Part

4 AppendixesThis part of the manual contains miscellaneous reference information in the following appendixes:• Appendix A, “Module and Specifier IDs,” on page 255• Appendix B, “SIP Response Codes,” on page 279• Appendix C, “Media Control Platform Reference Information,” on

page 289• Appendix D, “Default Device Profiles,” on page 304• Appendix E, “Specifications and Standards,” on page 307• Appendix F, “Caching Reference Information,” on page 315


User’s Guide 255

Appendix

A Module and Specifier IDsThis appendix lists various internal Genesys Voice Platform (GVP) identifiers that are required for advanced configuration of EMS Logging and Reporting.This appendix contains the following sections:

Media Control Platform, page 255Call Control Platform, page 267Resource Manager, page 271Fetching Module, page 276

For detailed information about the metrics (application-level logs) that the Media Control Platform and the Call Control Platform generate, including metric IDs and descriptions, see the Genesys Voice Platform 8.0 Metrics Reference.

Media Control PlatformTable 55 lists the Media Control Platform Application Module names and IDs.For the Next Generation Interpreter (NGI), see “Next Generation Interpreter Module ID and Specifiers” on page 265.

Table 55: Media Control Platform Application Module Names and IDs

Module Name Description or Comment Module ID Specifiers (link)

MTMPC Media Processing Component (MPC) wrapper

47 MTMPC

LMBase Base Line Manager 21 LMBase

LMSIP2 SIP Line Manager 40 LMSIP2

SESSMGR Call Manager API 28 SESSMGR


Appendix A: Module and Specifier IDs Media Control Platform

Table 56 lists the Media Control Platform specifier names and IDs.

CALLSESSION 29 None

SMMAIN Main module in the Media Control Platform 31 SMMAIN

CMUTIL Media Control Platform utility components 33 CMUTIL

APPMODULE Base Application Module 34 APPMODULE

REMDIAL Remote Dial Remdial Application Module 38 REMDIAL

CONFERENCE Conference Application Module 41 CONFERENCE

SQA Any and all SQA logs 43 SQA

MEDIAMGR The Media Manager part of the MPC 176 MEDIAMGR

CONTROL The control layer part of the MPC 177 CONTROL

MEDIA The media layer part of the MPC 178 MEDIA

RTP_INTERFACE The RTP layer of the MPC 179 RTP_INTERFACE

DSP The DSP components 180 DSP

VGULOGMOD_MAIN

The main Utility 128 VGULOGMOD_MAIN

MTINTERNAL The Internal Media Transport application module

130 MTINTERNAL

RTSPSTACK The RTSP Stack 132 RTSPSTACK

Table 55: Media Control Platform Application Module Names and IDs (Continued)

Module Name Description or Comment Module ID Specifiers (link)

Table 56: Media Control Platform Specifier Names and IDs

Specifier ID Specifier Name

MTMPC

1001 CMLOGMOD_MTMPC_INITFAILED

2001 CMLOGMOD_MTMPC_CONNERROR

LMBase

1001 CMLOGMOD_LMBASE_IDGENDIRUNACCBLE

User’s Guide 257


1003 CMLOGMOD_LMBASE_SYSIPNOTRETRVABLE

1004 CMLOGMOD_LMBASE_FAILUPDTEOPENCALLIDFILE

1005 CMLOGMOD_LMBASE_NOTPUTSEQNUMTOCALLIDFILE

2001 CMLOGMOD_LMBASE_RESETCALLIDFILECONTNTINVD

3001 CMLOGMOD_LMBASE_NOMEDIASESSPLAYAUDIO

3002 CMLOGMOD_LMBASE_NOMEDIASESSPLAYDTMF

3004 CMLOGMOD_LMBASE_NOMEDIASESSRECRDAUDIO

3005 CMLOGMOD_LMBASE_NOMEDIASESSSTREAMING

LMSIP2

2001 CMLOGMOD_LMSIP2_RECVUNEXPCTACK

2002 CMLOGMOD_LMSIP2_MEDIAERROR

2003 CMLOGMOD_LMSIP2_ERRSNDINVRESPONSE

2004 CMLOGMOD_LMSIP2_REGISTERTIMEOUT

2005 CMLOGMOD_LMSIP2_REGISTERBADREQUEST

2006 CMLOGMOD_LMSIP2_REGISTERFORBIDDEN

2007 CMLOGMOD_LMSIP2_REGISTERNOTFOUND

2008 CMLOGMOD_LMSIP2_REGISTERNOTACCEPTABLE

2009 CMLOGMOD_LMSIP2_REGISTEROTHERERROR

2010 CMLOGMOD_LMSIP2_VGSIPERRORNOTIFY

2011 CMLOGMOD_LMSIP2_ERRPARSESDPCONTENT

2012 CMLOGMOD_LMSIP2_REGISTERALGONOTSUPPORTED

2013 CMLOGMOD_LMSIP2_REGISTERAUTHENTICATIONERROR

2014 CMLOGMOD_LMSIP2_NONMATCHINGSIPINFO

2015 CMLOGMOD_LMSIP2_CUSTOMPARAMERROR

3001 CMLOGMOD_LMSIP2_CANTACCEPTNONINVITECALL

Table 56: Media Control Platform Specifier Names and IDs (Continued)




3002 CMLOGMOD_LMSIP2_ERRSNDINVITE

3003 CMLOGMOD_LMSIP2_ERRCREATERTPSESS

3004 CMLOGMOD_LMSIP2_ERRCREATEPSTNSESS

3005 CMLOGMOD_LMSIP2_BADDYNAMICPAYLOAD

3006 CMLOGMOD_LMSIP2_BADDTMFRECV

3007 CMLOGMOD_LMSIP2_ZEROCLOCKRATE

4001 CMLOGMOD_LMSIP2_MESSAGE

4002 CMLOGMOD_LMSIP2_PROCDELAY

SESSMGR

1001 CMLOGMOD_SESSMGR_IDGENDIRUNACCBLE

1003 CMLOGMOD_SESSMGR_SYSIPNOTRETRVABLE

1004 CMLOGMOD_SESSMGR_FAILUPDTEOPENCALLIDFILE

1005 CMLOGMOD_SESSMGR_NOTPUTSEQNUMTOCALLIDFILE

1007 CMLOGMOD_SESSMGR_VRMINITFAIL

1008 CMLOGMOD_SESSMGR_CANTINITLICENSEMGR

2001 CMLOGMOD_SESSMGR_ATTEMPTAUDIOCTRLWBARGEIN

2002 CMLOGMOD_SESSMGR_BADFRMTSCPTAUDIO

2003 CMLOGMOD_SESSMGR_BADFRMTSCPTTTS

2004 CMLOGMOD_SESSMGR_BADFRMTSCPTSTRMNG

2016 CMLOGMOD_SESSMGR_OUTCALLNORESOURCE

2017 CMLOGMOD_SESSMGR_TTSMGRLOST

2018 CMLOGMOD_SESSMGR_TRFRTODESTNOTAUTH

2019 CMLOGMOD_SESSMGR_DESTURINOTSUPP

2020 CMLOGMOD_SESSMGR_DESTURIMALFORMED

2021 CMLOGMOD_SESSMGR_STRMMODUNEXPTEVENT



User’s Guide 259


2022 CMLOGMOD_SESSMGR_LOSTASRMGR

2023 CMLOGMOD_SESSMGR_INITCALLSESSWNOLNMGR

2024 CMLOGMOD_SESSMGR_RESETCALLIDFILECONTNTINVD

2026 CMLOGMOD_SESSMGR_ISDNCAUSECODEERR

3001 CMLOGMOD_SESSMGR_UNEXPECTTTSERROR

3002 CMLOGMOD_SESSMGR_EXPIREASRTTSIGNORED

3003 CMLOGMOD_SESSMGR_UNEXPECTCMCALLBILLEVENT

3005 CMLOGMOD_SESSMGR_FAILMEDSTRMRESLT

3006 CMLOGMOD_SESSMGR_APPMODULENOTFOUND

4001 CMLOGMOD_SESSMGR_INBOUNDDTMF

4003 CMLOGMOD_SESSMGR_NOINOUTLINES

SMMAIN

1001 CMLOGMOD_SMMAIN_VRMDLLLOADFAIL

1002 CMLOGMOD_SMMAIN_VRMSETLOGFAIL

1003 CMLOGMOD_SMMAIN_MAKEVRMFAIL

1004 CMLOGMOD_SMMAIN_CREATEVRMFAIL

1006 CMLOGMOD_SMMAIN_CALLMGRCFGPARAMERR

1007 CMLOGMOD_SMMAIN_LOADTOOMANYCMGRMOD

1008 CMLOGMOD_SMMAIN_FAILCREATECMGRMOD

1009 CMLOGMOD_SMMAIN_LOADTOOMANYDEVICE

1010 CMLOGMOD_SMMAIN_FAILCREATEDEVICE

1011 CMLOGMOD_SMMAIN_FAILINITDEVICE

1012 CMLOGMOD_SMMAIN_LOADTOOMANYMEDTRPT

1013 CMLOGMOD_SMMAIN_FAILCREATEMEDTRPT

1014 CMLOGMOD_SMMAIN_FAILINITMEDTRPT





1015 CMLOGMOD_SMMAIN_LOADTOOMANYLNMGRS

1016 CMLOGMOD_SMMAIN_FAILCREATELNMGR

1017 CMLOGMOD_SMMAIN_FAILINITLNMGR

1018 CMLOGMOD_SMMAIN_SESSMGRAPPMODCFGERR

1019 CMLOGMOD_SMMAIN_LOADTOOMANYAPPMOD

1020 CMLOGMOD_SMMAIN_SESSMGRMODCFGERR

1021 CMLOGMOD_SMMAIN_LOADTOOMANYSESSMOD

1022 CMLOGMOD_SMMAIN_FAILOPENLICENSE

1023 CMLOGMOD_SMMAIN_FAILPARSELICENSE

1024 CMLOGMOD_SMMAIN_MACVALIDERR

1025 CMLOGMOD_SMMAIN_GENINITLICERR

1026 CMLOGMOD_SMMAIN_CANTCREATEVGNETLIB

1027 CMLOGMOD_SMMAIN_CANTINITVGNETLIB

1028 CMLOGMOD_SMMAIN_FAILINITCFGOBJ

1029 CMLOGMOD_SMMAIN_CANTSTARTCMGR

2002 CMLOGMOD_SMMAIN_FAILLOADAPPMODLIB

2003 CMLOGMOD_SMMAIN_FAILINITAPPMOD

2004 CMLOGMOD_SMMAIN_NOVLDAPPMODINLIB

2005 CMLOGMOD_SMMAIN_LIBNODEFMAKEAPPMOD

2006 CMLOGMOD_SMMAIN_VXMLAPPMODNOTLOAD

CMUTIL

2001 CMLOGMOD_CMUTIL_TELNUMLONG

2002 CMLOGMOD_CMUTIL_TELNUMINVCHAR

2003 CMLOGMOD_CMUTIL_POSTDIALLONG

2004 CMLOGMOD_CMUTIL_POSTDIALINVCHAR



User’s Guide 261


2005 CMLOGMOD_CMUTIL_CONFLICTEXT

2006 CMLOGMOD_CMUTIL_HUNTGPINVTRUNK

3001 CMLOGMOD_CMUTIL_HUNTGPNONEXISTTRUNK

3002 CMLOGMOD_CMUTIL_CALLREQNONEXISTHUNTGP

3003 CMLOGMOD_CMUTIL_WAITFORDIAL

3004 CMLOGMOD_CMUTIL_ATTRIBLONG

3005 CMLOGMOD_CMUTIL_VALUELONG

APPMODULE

1001 CMLOGMOD_APPMODULE_FAILSTRTWORKNGTHRD

2001 CMLOGMOD_APPMODULE_FAILREGAPP

2002 CMLOGMOD_APPMODULE_FAILREGAPPMOD

2003 CMLOGMOD_APPMODULE_FAILBINDAPP

REMDIAL

2001 CMLOGMOD_REMDIAL_FAILREGREMDLMOD

2002 CMLOGMOD_REMDIAL_CANTCREATESERVERSOCK

2003 CMLOGMOD_REMDIAL_SOCKETERROR

3001 CMLOGMOD_REMDIAL_MAXCALLSWARN

3002 CMLOGMOD_REMDIAL_MAXCLIENTS

3003 CMLOGMOD_REMDIAL_NOACTIVESESS

3004 CMLOGMOD_REMDIAL_MAXCALLSREACHED

CONFERENCE

2001 CMLOGMOD_CONFERENCE_FAILED

2002 CMLOGMOD_CONFERENCE_UNEXPTREASON

4001 CMLOGMOD_CONFERENCE_ESTABLISHED

4002 CMLOGMOD_CONFERENCE_TERMINATED





SQA

4001 CMLOGMOD_SQA_DTMF

4002 CMLOGMOD_SQA_TRANSFERSTART

4003 CMLOGMOD_SQA_TRANSFEREND

4004 CMLOGMOD_SQA_PROMPTTYPE

4006 CMLOGMOD_SQA_RECOGNITIONSTART

4007 CMLOGMOD_SQA_RECOGNITIONEND

4008 CMLOGMOD_SQA_OPENRECORDFILE

4009 CMLOGMOD_SQA_ClOSERECORDFILE

4010 CMLOGMOD_SQA_MEDIAROUTING

4011 CMLOGMOD_SQA_AUDIOGAP

4012 CMLOGMOD_SQA_FIRSTAUDIOPK

4013 CMLOGMOD_SQA_LASTAUDIOPK

427820 CMLOGMOD_SQA_ECMASCRIPT_TIMINGS

427801 CMLOGMOD_SQA_COMPILE_TIME

427802 CMLOGMOD_SQA_FETCH_TIME

MEDIAMGR

2001 MPCLOGMOD_MEDIAMGR_INVALIDMEDIA

2002 MPCLOGMOD_MEDIAMGR_UNEXPECTEDRTSPDISC

2003 MPCLOGMOD_MEDIAMGR_RTSPREQFAIL

2004 MPCLOGMOD_MEDIAMGR_RTSPREPLYERROR

2005 MPCLOGMOD_MEDIAMGR_RTSPRTPERROR

2006 MPCLOGMOD_MEDIAMGR_UNSUPPORTEDVIDFMT

2008 MPCLOGMOD_MEDIAMGR_UNSUPPORTEDAUDCHNLS

2009 MPCLOGMOD_MEDIAMGR_BADAVICHNKSIZE



User’s Guide 263


2010 MPCLOGMOD_MEDIAMGR_MALFORMEDAVIHDR

2011 MPCLOGMOD_MEDIAMGR_RECBUFFISOTOOSMALL

2012 MPCLOGMOD_MEDIAMGR_UNABLETOALLOCMEM

2013 MPCLOGMOD_MEDIAMGR_NOISOTRAK

2014 MPCLOGMOD_MEDIAMGR_BADISOBOXSIZE

2016 MPCLOGMOD_MEDIAMGR_BRANDINCOMPT3GPP

2017 MPCLOGMOD_MEDIAMGR_BADMAJ3GPPBRAND

2018 MPCLOGMOD_MEDIAMGR_ERORISOBOXVALUE

2019 MPCLOGMOD_MEDIAMGR_FAILTOSTARTRECORD

2020 MPCLOGMOD_MEDIAMGR_NOMEDIAINFOOBJECT

3001 MPCLOGMOD_MEDIAMGR_RECFRAMEDISCARD

3002 MPCLOGMOD_MEDIAMGR_UNEXPECTEDRTSPREPLY

3003 MPCLOGMOD_MEDIAMGR_BADISOBOXVALUE

3004 MPCLOGMOD_MEDIAMGR_BADISOBOXTYPE

3005 MPCLOGMOD_MEDIAMGR_MANDISOBOXMISS

3006 MPCLOGMOD_MEDIAMGR_BUFFTOOSMALLTOPARSEISOHDR

3007 MPCLOGMOD_MEDIAMGR_UNSUPPORTEDAUDRATE

3008 MPCLOGMOD_MEDIAMGR_UNSUPPORTEDVIDRATE

CONTROL

1001 MPCLOGMOD_CONTROL_INITVGMEDIAINFOFAILED

1002 MPCLOGMOD_CONTROL_INITDSPCAPFAILED

2001 MPCLOGMOD_CONTROL_INVALIDHRTIMERRES

2002 MPCLOGMOD_CONTROL_SDPPARSEFAILED

3001 MPCLOGMOD_CONTROL_INVALIDCFG

3002 MPCLOGMOD_CONTROL_CONNINITFAILED





3003 MPCLOGMOD_CONTROL_CONNMODIFYFAILED

3004 MPCLOGMOD_CONTROL_SENDDTMFNOTALLOWED

3005 MPCLOGMOD_CONTROL_INVALIDCONFIGPARAM

MEDIA

2001 MPCLOGMOD_MEDIA_RECORDOPENFAILED

3001 MPCLOGMOD_MEDIA_ACCESSFAILED

3002 MPCLOGMOD_MEDIA_SINKBUFFERFULL

3003 MPCLOGMOD_MEDIA_SOURCEBUFFERFULL

3004 MPCLOGMOD_MEDIA_PACKETBUFFERFULL

3005 MPCLOGMOD_MEDIA_RTPPACKETTOOLARGE

3006 MPCLOGMOD_MEDIA_BUFFERTOOSMALL

3007 MPCLOGMOD_MEDIA_BRIDGEOBJECTNOTFOUND

3008 MPCLOGMOD_MEDIA_H263SORTEROUTOFPACKET

3009 MPCLOGMOD_MEDIA_SILENCEFILLDISABLED

3010 MPCLOGMOD_MEDIA_SENDDTMFDISABLED

3011 MPCLOGMOD_MEDIA_NORTPSTREAMSENDDTMF

3012 MPCLOGMOD_MEDIA_NORTPSTREAMMEDIATRANSMIT

3013 MPCLOGMOD_MEDIA_ERRORDECODINGRFC2833

RTP_INTERFACE

3001 MPCLOGMOD_RTPIF_INCORRECTTIMEINDEX

3002 MPCLOGMOD_RTPIF_OUTOFSEQUENCEINCOMINGRTP

3003 MPCLOGMOD_RTPIF_INCOMINGRTPDELAY

3004 MPCLOGMOD_RTPIF_ERRORDEFRAMINGPACKET

3005 MPCLOGMOD_RTPIF_UNEXPECTEDPAYLOADTYPE



User’s Guide 265


Next Generation Interpreter Module ID and SpecifiersThe Module ID for the Next Generation Interpreter (NGI) application is 192.Table 57 describes the specifiers for the NGI application module.

DSP

3002 MPCLOGMOD_DSP_NOTRANSCODER

VGULOGMOD_MAIN

1002 VGLOG_CANT_OPEN_DLL

2003 VGLOG_SOCKET_SEND_FAILED

7001 VGLOG_TRACE_GENERIC

MTINTERNAL

2001 VGLOG_MTINTERNAL_MINORMAXPORT

2002 VGLOG_MTINTERNAL_MINLARGERTHANMAX

3001 VGLOG_MTINTERNAL_OPENFILEERROR

3002 VGLOG_MTINTERNAL_SENDDATAERROR

3003 VGLOG_MTINTERNAL_WRITEFILEERROR

3004 VGLOG_MTINTERNAL_DISCARDRTPPACKET

RTSPSTACK

2001 VGLOG_RTSP_NEW_FAILED

2002 VGLOG_RTSP_INVALID_CONFIG

2003 VGLOG_RTSP_UNINIT

2004 VGLOG_RTSP_CONSTRUCT_BAD_MSG

2005 VGLOG_RTSP_PARSE_BAD_MSG

2006 VGLOG_RTSP_SOCKET_ERROR

3001 VGLOG_RTSP_SOCKET_EVENT

5001 VGLOG_RTSP_SOCKET_CLOSE





Table 57: NGI Specifiers

Specifier ID Specifier Name Level

3501 NGI_LOG_JS_WARNING Warning

3502 NGI_LOG_JS_INFO Info

3503 NGI_LOG_NET_CONNECT_FAILURE Warning

3504 NGI_LOG_NET_CONNECT_FAILURE_INFO Info

3505 NGI_LOG_INITIALIZE_ERROR Warning

3506 NGI_LOG_CREATE_DIALOG_FAILURE Info

3507 NGI_LOG_CONFIGURATION Info

3508 NGI_LOG_INVALID_PROPERTY Info

3509 NGI_LOG_INVALID_SYNTAX Warning

3510 NGI_LOG_UNEXPECTED_WARNING Warning

1000 NGI_LOG_CONVERSION Info

1001 NGI_LOG_CONVERSION_WARNING Warning

1002 NGI_LOG_APPLICATION_ERROR Info (vxml application)

1003 NGI_LOG_APPLICATION_WARNING Warning (vxml application)

1004 NGI_LOG_SOMETHING_UNEXPECTED Error

1005 NGI_LOG_UNEXPECTED_WARNING Warning

1006 NGI_LOG_INCALL_SETUP_FAILURE Error

1007 NGI_LOG_CREATE_CALL_FAILURE Error

1008 NGI_LOG_NGI_INITIALIZATION_FAILURE Error

1009 NGI_LOG_CONFIGURATION_WARNING Warning

1010 NGI_LOG_RECORDED_FILE_TOO_SMALL Warning

1011 NGI_LOG_FETCH_FAILURE Info

1012 NGI_LOG_FETCH_FAILURE_WARNING Warning

1013 NGI_LOG_GRAMMAR_ERROR Info

User’s Guide 267

Appendix A: Module and Specifier IDs Call Control Platform

Call Control PlatformTable 58 lists the Call Control Platform Application Module names and IDs.

Connection, Dialog, or Conference EventsTable 59 describes the specifiers for the main Call Control Platform module (Module ID = 151). These events are related to a connection, dialog, or conference.

1014 NGI_LOG_PROMPT_FETCH_TIMEOUT Error

1015 NGI_LOG_PROMPT_FETCH_ERROR Error

Table 57: NGI Specifiers (Continued)

Specifier ID Specifier Name Level

Table 58: Call Control Platform Application Module Names and IDs

Module Module ID

Main Call Control Platform (CCP) application module. 151

CCXML Interpreter 152

Media Controller 153



Table 59: CCP Connection, Dialog, or Conference Events

Module ID Specifier ID Description

151 Critical Events

1 Failed to initialize software.

Error Events


257 Inbound connection failure.

258 Media Controller reported error.

Warning Events

514 Inbound connection rejected while in suspended state.

515 Application did not specify an event name for <send>.

516 History Info header is malformed.

517 Invalid hints passed.

151 (continued)

Info Events

1025 Connection created.

1026 Connection terminated.

1027 Sending 180 Ringing automatically as configured.

1044 Conference created.

1045 Conference terminated.

1046 Dialog created.

1047 Dialog terminated.

1048 Dialog transfer request rejected per configured.

1049 Buffering up join request until ready.

1050 Issuing buffered join requests.

User’s Guide 269


Media Controller EventsTable 60 describes the Media Controller events.

Table 60: CCP Media Controller Events


153 Critical Events


Error Events


257 Device profile entry empty.

258 Inbound call leg offer was rejected by application.

259 Bridging server encountered error.

260 Failure to initialize the Session Factory.

Warning Events

514 Uninitialization encountered problems.

515 Operation issued on the leg failed.

516 SDP generation/processing encountered problems.

517 SIP 491 Glare occurred.

518 Maximum number of retries reached.

519 Maximum number of updates reached.

520 Operation execution failed.

521 NULL Operation added to Transaction.

522 CallTerminate received and state is either DISCONNECTED or ERROR.

Info Events

1024 Connection timeout.

1025 Dialog Unsupported MIME Type.

1026 Conference created.



153 (continued)

1027 Conference terminated.

1028 Standard Conference creation.

1029 Implicit Conference creation.

1030 Media established.

1031 Media modified.

1032 Media terminated.

1033 SIP-CallID: [<SIP Call-ID>]; SendInvite() Failed[<returncode>].

1034 SIP-CallID: [<SIP Call-ID>]; SendResponse(<SIP code>) for <SIP method> Failed[<returncode>].

1035 SIP-CallID: [<SIP Call-ID>]; SendCancel() Failed[<returncode>].

1036 SIP-CallID: [<SIP Call-ID>]; SendRequest(<SIP method>) Failed[<returncode>].

1037 SIP-CallID: [<SIP Call-ID>]; SendInfo() Failed[<returncode>].

1038 SIP-CallID: [<SIP Call-ID>]; SendBye() Failed[<returncode>].

1039 SIP-CallID: [<SIP Call-ID>]; SendAck() Failed[<returncode>].

1040 Device profile selected.

1041 List of operations in the transaction: <OP1, OP2, ...>.

1042 SIP UserCall not connected.

1043 SIP UserCall error state.

1044 SIP UserCall received a failure response.

1045 SIP UserCall failed sending a message.

Table 60: CCP Media Controller Events (Continued)


User’s Guide 271

Appendix A: Module and Specifier IDs Resource Manager

Log_4 (INFO) EventsTable 61 describes the CCXML interpreter events at the INFO level.

Resource ManagerThe Module ID for the Resource Manager application is 148.Table 62 describes the specifiers for the Resource Manager application module.

Table 61: CCXMLI Log_4 INFO Events


152 1024 CCXMLI initialized.

1025 CCXMLI uninitialized.

1026 A new CCXML session created.

1027 A CCXML session terminated.

1028 Failed to fetch document.

1029 Failed to parse document.

1030 Failed to compile document.

1031 Document initialization failed.

1032 Event not caught by application.

1033 Application log (by <log> tag).

1034 Error event generated by application.

1035 Exceed maximum session limit.

Table 62: Resource Manager Specifiers


257 GVPLOG_RM_UNRECOVERABLEERR

513 GVPLOG_RM_CONFIGERR

514 GVPLOG_RM_CCPSS7ERR

515 GVPLOG_RM_SOCKETERR



516 GVPLOG_RM_RESOURCEALLOCERR

517 GVPLOG_RM_CDRINITERR

518 GVPLOG_RM_CDRUNINITERR

519 GVPLOG_RM_CDRRECORDCREATEERR

520 GVPLOG_RM_CDRRECORDDELETEERR

521 GVPLOG_RM_DIALINGRANGEEXCEED

522 GVPLOG_RM_DIALINGTYPEINVALID

523 GVPLOG_RM_DIALINGEXPRINVALID

524 GVPLOG_RM_DNISNOTEXIST

525 GVPLOG_RM_DEFAULTTENTANTNOTFOUND

526 GVPLOG_RM_REQUESTURITRANSLATIONFAIL

527 GVPLOG_RM_CALLCREATEFAIL

528 GVPLOG_RM_APPPROFILENOTFOUND

529 GVPLOG_RM_TENANTNOTFOUND

530 GVPLOG_RM_DEFAULTIVRPROFILENOTFOUND

531 GVPLOG_RM_DEFAULTSERVICETYPENOTFOUND

532 GVPLOG_RM_MANDATORYURIPARAMNOTFOUND

533 GVPLOG_RM_INVALIDURIPARAM

534 GVPLOG_RM_SERVICEPREREQNOTFOUND

535 GVPLOG_RM_NOMATCHINGSERVICETYPE

536 GVPLOG_RM_NOMATCHINGGWPREFERENCE

537 GVPLOG_RM_CCILIBINVALIDPARAM

538 GVPLOG_RM_CCILIBCONFIGOBJERR

539 GVPLOG_RM_CCILIBRMOBJERR

540 GVPLOG_RM_CCILIBRESOBJNOTFOUND

Table 62: Resource Manager Specifiers (Continued)


User’s Guide 273


541 GVPLOG_RM_CCILIBLOGICALRESCREATEFAIL

542 GVPLOG_RM_CCILIBPHYSICALRESCREATEFAIL

543 GVPLOG_RM_CCILIBTENANTNOTFOUND

544 GVPLOG_RM_CCILIBTENANTCREATEFAIL

545 GVPLOG_RM_CCILIBAPPIDNOTFOUND

546 GVPLOG_RM_CCILIBLINKEDRESNOTFOUND

547 GVPLOG_RM_CCILIBPARENTNOTFOUND

548 GVPLOG_RM_CCILIBLOGICALRESGROUPNOTFOUND

549 GVPLOG_RM_CCILIBTENANTCONVERTERROR

550 GVPLOG_RM_CCILIBCAPADDERROR

551 GVPLOG_RM_CCILIBAPPCONVERTERROR

552 GVPLOG_RM_CCILIBINVALIDINPUTARG

553 GVPLOG_RM_RESSESSIONCREATEFAIL

554 GVPLOG_RM_CCILIBAPPCREATEFAIL

555 GVPLOG_RM_CCILIBUPDATEINVALIDCFGOBJ

556 GVPLOG_RM_CCILIBUPDATETENANTNOTFOUND

557 GVPLOG_RM_CCILIBUPDATETENANTPOPULATEFAIL

558 GVPLOG_RM_CCILIBUPDATEAPPNOTFOUND

559 GVPLOG_RM_CCILIBUPDATEAPPPOPULATEFAIL

560 GVPLOG_RM_CCILIBUPDATELOGICALRESNOTFOUND

561 GVPLOG_RM_CCILIBUPDATELOGICALRESADDERR

562 GVPLOG_RM_CCILIBUPDATERESOBJNOTFOUND

563 GVPLOG_RM_CCILIBUPDATEPHYRESCREATEFAIL

564 GVPLOG_RM_CCILIBUPDATEINVALIDOBJ

565 GVPLOG_RM_CCILIBUPDATEINVALIDOBJTYPE





566 GVPLOG_RM_CCILIBUPDATETENANTADDFAIL

567 GVPLOG_RM_CCILIBUPDATEAPPADDFAIL

568 GVPLOG_RM_CCILIBUPDATETENANTREMOVEFAIL

569 GVPLOG_RM_CCILIBUPDATEAPPREMOVEFAIL

570 GVPLOG_RM_CCILIBUPDATETENANTUPDATEFAIL

571 GVPLOG_RM_CCILIBUPDATEAPPLICATIONUPDATEFAIL

572 GVPLOG_RM_REGISTERERROR

574 GVPLOG_RM_POLICYVIOLATIONERROR

769 GVPLOG_RM_INVALIDMSG

770 GVPLOG_RM_INVALIDCONFIG

771 GVPLOG_RM_CCPSS7SUBSERFAIL

772 GVPLOG_RM_NETWORKPROBLEM

773 GVPLOG_RM_REQUESTURIPARSEFAIL

774 GVPLOG_RM_OPTIONUSERINFOEXIST

775 GVPLOG_RM_TOHEADERPARSEFAIL

776 GVPLOG_RM_RMSERVICEAGENTBADMSGFORMAT

777 GVPLOG_RM_RMSUSPEND

778 GVPLOG_RM_SIPSERVICESAMEPRECEDENCE

779 GVPLOG_RM_INVALIDCALLTENANTID

780 GVPLOG_RM_FAILEDTOFINDLINKEDTENANT

781 GVPLOG_RM_FAILEDTOFINDLINKEDRESOURCE

782 GVPLOG_RM_LOGICALRESINFONOTFOUND

783 GVPLOG_RM_LOGICALRESPOPULATEFAIL

784 GVPLOG_RM_LOGICALRESSECTIONNOTFOUND

785 GVPLOG_RM_PHYSRESPOPULATEFAIL



User’s Guide 275


786 GVPLOG_RM_TENANTPOPULATEINCOMPLETE

787 GVPLOG_RM_APPINFONOTFOUND

788 GVPLOG_RM_APPPOPULATEINCOMPLETE

789 GVPLOG_RM_DNISEXTRACTFAIL

790 GVPLOG_RM_SETTINGLOGICALRESPROPERTIES

791 GVPLOG_RM_AORNOTFOUND

792 GVPLOG_RM_CAPACITYNOTFOUND

793 GVPLOG_RM_CAPACITYNONUNSIGNED

794 GVPLOG_RM_SETTINGPHYRESPROPERTIES

795 GVPLOG_RM_UPDATELOGICALRESGROUPNOTFOUND

796 GVPLOG_RM_UPDATEPOPULATEPHYRESFAIL

797 GVPLOG_RM_UPDATEFAILGETPHYRES

798 GVPLOG_RM_UPDATEPOPULATELOGICALRESFAIL

799 GVPLOG_RM_UPDATELOGICALRESNOTFOUND

800 GVPLOG_RM_UPDATEPHYRESREMOVED

801 GVPLOG_RM_UPDATETENANTADDED

802 GVPLOG_RM_UPDATEAPPADDED

803 GVPLOG_RM_UPDATELINKEDTENANTREMOVED

804 GVPLOG_RM_UPDATEAPPREMOVED

805 GVPLOG_RM_UDPATELINKEDTENANTUPDATED

806 GVPLOG_RM_UDPATEAPPDATAUPDATED

807 GVPLOG_RM_UPDATEIGNORED

808 GVPLOG_RM_WARNING_BAD_REGEX

1025 GVPLOG_RM_CCPSS7STATE

1026 GVPLOG_RM_COMMNOTICE




Appendix A: Module and Specifier IDs Fetching Module

Fetching ModuleThe Module ID for the Fetching Module application is 80.Table 63 describes the specifiers for the Fetching Module application module.

1027 GVPLOG_RM_CLUSTERNOTICE

1028 GVPLOG_RM_CCPPROXYSTATE

1029 GVPLOG_RM_STARTUP

1030 GVPLOG_RM_SHUTDOWN

1031 GVPLOG_RM_RMSERVICEAGENTSTATUS

1281 GVPLOG_RM_PROVCHANGE

1282 GVPLOG_RM_CCPSS7NOTIFY

1283 GVPLOG_RM_MODULECONNECTIVITY

1284 GVPLOG_RM_MODULECONFIGMODIF

1285 GVPLOG_RM_CLUSTERINFO

1286 GVPLOG_RM_NEWCALL

1287 GVPLOG_RM_REGISTERINFO

2003 VGLOG_SOCKET_SEND_FAILED

2305 GVPLOG_RM_GENERIC_TRACE



Table 63: Fetching Module Specifiers

Specifier ID Specifier Name Description

Level: Critical

40000 FMLOG_MEM_ALLOC_FAIL Memory allocation failed for %s.

40001 FMLOG_FM_INIT_FAIL Fetching Module initialization failed.

Level: Error

20020 FMLOG_SESS_OPEN_FAIL Open Session to Fetching Server failed.

User’s Guide 277


20021 FMLOG_CONN_FAIL Connect to Fetching Server failed.

20022 FMLOG_SEND_FAIL Send to Fetching Server failed.

20023 FMLOG_BAD_SESS_ID Invalid session ID.

20006 FMLOG_EMSLOG_INIT_FAIL EMS logging service initialization failed.

20007 FMLOG_BAD_SHMEM_PARAM Invalid shared memory parameter.

20008 FMLOG_SHMEM_NAME_EMPTY Empty shared memory name.

20009 FMLOG_SHSEM_NAME_FAIL Shared semaphore name generation failed.

20010 FMLOG_SHSEM_CREATE_FAIL Shared semaphore creation failed.

20011 FMLOG_SHSEM_LOCK_FAIL Shared semaphore lock failed.

20012 FMLOG_SHMEM_MAP_FAIL Shared memory map failed for file %s.

20013 FMLOG_SHMEM_ATTACH_FAIL Shared memory attach failed for ID %d.

20014 FMLOG_SHMEM_NAME_FAIL Shared memory name generation failed.

20015 FMLOG_SHMEM_CREATE_FAIL Shared memory creation failed for size %d.

20016 FMLOG_SHMEM_READ_FAIL Unable to read shared-memory.

20017 FMLOG_SHMEM_WRITE_FAIL Unable to write shared-memory.

20018 FMLOG_GET_PIPE_FAIL Failed to get pipe name.

20019 FMLOG_OPEN_PIPE_FAIL Failed to open pipe.

Level: Warning

30000 FMLOG_SESS_CLS_FAIL Close Session to Fetching Server failed.

30003 FMLOG_CLS_PIPE_FAIL Failed to close pipe.

Table 63: Fetching Module Specifiers (Continued)

Specifier ID Specifier Name Description



User’s Guide 279

Appendix

B SIP Response CodesThis appendix lists the Session Initiation Protocol (SIP) responses that Genesys Voice Platform (GVP) components send or receive in response to error conditions and other events.It contains the following section:

SIP Responses within GVP, page 279For information about how the Media Control Platform handles error responses that it receives for outbound call requests, see Table 72 on page 299.

SIP Responses within GVPTable 64 summarizes the SIP response codes, other than the normal 200 OK responses, that the Resource Manager (RM), Media Control Platform (MCP), and Call Control Platform (CCP) signal in response to error conditions and other events during incoming and outbound call setup and processing. Table 64 also lists the configuration options to customize those responses, where applicable.The Resource Manager handles SIP responses from other components in accordance with rules described on page 36.

Table 64: SIP Response Codes

SIP Response Sent By Situations Configurable Options and Notes

Code Phrase

100 Trying MCPCCPRM*

The immediate response to a valid INVITE request.

*For CCP-initiated outbound calls.


Appendix B: SIP Response Codes SIP Responses within GVP

180 Ringing MCP The default intermediate response to an INVITE request.

sip.sendalert (see page 175)

CCP Intermediate response sent for all incoming calls, on <accept> (no media bridge configured). Depending on configuration:• Response is sent when <send>

is called.• Response is sent immediately

after sending 100 Trying.

ccpccxml.sip.send_progressing

183 Session Progress

MCP The non-default intermediate response, which includes SDP information.

sip.sendalert (see page 175)

MCP An incoming call is being offered to the Next Generation Interpreter (NGI) for debugging.The NGI passes the debugger IP address and port information to the calling party in the following SIP headers:• X-GVP-NGI-DEBUG-IP

• X-GVP-NGI-DEBUG-PORT

Note: The information can also be sent in INVITE messages.

Sent if debugging is enabled on the MCP:vxmli.debug.enabled

Table 64: SIP Response Codes (Continued)


Code Phrase

User’s Guide 281


183 Session Progress (continued)

CCP Intermediate response sent for incoming calls when a media bridge has been configured between this bridge and any other endpoint. The response includes the appropriate SDP content.

If media bridging changes are required, they are implemented through subsequent SDP updates in re-INVITE, 200 OK, and 183 messages.Note: When a BYE is received on a SIP dialog that is associated with an endpoint while a transition involving that endpoint is being executed, any new bridge involving the endpoint will fail. The error.connection.join event is thrown, with an empty Reason property.

202 Accepted MCP A REFER request to initiate an outbound call outside of a SIP dialog is accepted by the VoiceXML application.

3xx [Various] MCP The MCP, acting as a User Agent Server (UAS), failed to negotiate a media session or the NETANN request was malformed.

See Warning header information in Table 71 on page 295.

302 Moved Temporarily

CCP The platform is redirecting a call in the ALERTING state (<redirect> tag).If the CCXML application specifies a <reason> attribute, the reason is included in the text portion of the Reason header.

To customize the SIP response code for specific situations, use the <hints> attribute of the <redirect> tag—the responseCode property of the hints object specifies the response code to be used.

400 Bad Request RM Malformed Request-URI in a REGISTER message.

MCP Malformed Request-URI in an INVITE message.

CCP The initial CCXML page URI is malformed.



Code Phrase



403 Forbidden RM The domain name in a REGISTER request does not match the configured domain name.

RM Call is rejected because of an IVR Profile policy (dialing rule).

IVR Profile:gvp.policy.dialing-rule-forbidden-respcode

Note: An equivalent configuration option also enables an alarm to be set.

RM Service request is rejected because the IVR Profile policy does not allow the service in the session.

IVR Profile:gvp.policy.conference-forbidden-respcode

gvp.policy.external-sip-forbidden-respcode

gvp.policy.outbound-call-forbidden-respcode

gvp.policy.transfer-forbidden-respcode

gvp.policy.voicexml-dialog-forbidden-respcode

Note: Equivalent configuration options also enable an alarm to be set.

404 Not Found RM The Resource Manager could not match the incoming request to an IVR Profile.

RM The Resource Manager could not match the incoming request to a service.

MCP The Request-URI has conf as the user part, but does not have a conf-id parameter.

405 Method Not Allowed

RM REFER, OPTIONS, SUBSCRIBE, or INFO message was sent outside of an existing SIP dialog.



Code Phrase

User’s Guide 283


408 Request Timeout

RM The Resource Manager does not receive a response from the resource.

RM The Resource Manager has not received a response from any resource.

RM An INVITE request specifies a host for which the Resource Manager is not responsible. (By default, no responsible domains are specified, so all requests are accepted.)

420 Bad Extension

MCP The MCP rejects an incoming call that requires 100rel (SIP Provisional Message Reliability).

423 Interval Too Brief

RM The Resource Manager received a REGISTER request for a registration period that fell below the configured minimum expiry time.

480 Temporarily Unavailable

RM Conference call fails because of insufficient resource port capacity or because conference has reached maximum size.

rm.conference-sip-error-respcode

RM The Resource Manager is not able to select a resource to which to forward the request, because a suitable resource is not available.

rm.resource-unavailable-respcode

RM The Resource Manager is not able to select a resource to which to forward the request, because the deployment does not include the required resource.

rm.resource-no-match-respcode



Code Phrase



480 Temporarily Unavailable (continued)

RM Call is rejected because usage limits, as specified in the IVR Profile policy, have been exceeded.

IVR Profile:gvp.policy.usage-limit-exceeded-respcode


CCP The platform is rejecting an incoming connection in the ALERTING state (<reject> tag).If the CCXML application specifies a <reason> attribute, the reason is included in the text portion of the Reason header.

ccpccxml.defaultrejectcode

To further customize the SIP response code for specific situations, use the <hints> attribute of the <reject> tag—the responseCode property of the hints object specifies the response code to be used.

CCP The platform is not in READY state, and is therefore rejecting all INVITE and OPTIONS requests.


487 Request Terminated

MCP A CANCEL or BYE is received before the final response to the INVITE was sent.

488 Not Acceptable Here

CCP An endpoint’s SDP capabilities cannot be obtained.

The response includes a Warning header with warning code 399, and warning text one of the following:• Unable to generate an offer—The INVITE contained no SDP.

• Unable to generate an answer—All other situations.



Code Phrase

User’s Guide 285


500 Server Internal Error

RM The Resource Manager received a REGISTER from a resource about which it had no information from Management Framework.

RM The Resource Manager does not receive a 2xx response from any resource (see page 36).

MCP Unable to create a media session to handle the call.

The response includes an explanatory Warning header (see Warning header information in Table 71 on page 295).

MCP The MCP is unable to fetch or parse the VoiceXML document.

503 Service Unavailable

RM The Resource Manager is in suspend mode when a new session request arrives.

rm.suspend-mode-respcode



Code Phrase



503 Service Unavailable(continued)

RM Call is rejected because usage limits for the service, as specified in the IVR Profile policy, have been exceeded.

IVR Profile:gvp.policy.ccxml-usage-limit-exceeded-respcode

gvp.policy.conference-usage-limit-exceeded-respcode

gvp.policy.voicexml-usage-limit-exceeded-respcode


MCP A conference cannot be created because of a resource problem (for example, failed to join to conference because of the conference limit).

MCP The Media Server is not accepting new calls for a reason other than those covered by the 500 response.

All error responses to an INVITE request, if they do not involve SDP negotiation, will contain a Warning header with a value of 399 and a human-readable description of the error.

MCP The VoiceXML application could not be fetched or parsed.

CCP The platform is not in READY state, and is therefore rejecting all HTTP requests to start a new CCXML session.


BYE message CCP Bridge failure resulting from the failure of endpoints to negotiate SDP might cause the CCP to send SIP BYE messages to the components involved.

The Reason header value is set to Application Disconnect.

[Configurable] RM The Resource Manager returns a response to a SIP OPTIONS message

rm.options_response_contenttype

rm.options_response_msg_body



Code Phrase

User’s Guide 287


[Configurable] SIP Server When the Resource Manager receives any of the configured SIP responses from SIP Server for a request to a gateway resource, it will retry the request on other gateway resources in the logical resource group.

Resource Manager:<gateway resource group>.noresource-response-code



Code Phrase



User’s Guide 289

Appendix

C Media Control Platform Reference InformationThis appendix provides miscellaneous reference information about the Media Control Platform.It contains the following sections:

Audio and Video File Formats, page 289Dynamic Media Control Platform Parameters, page 294SIP Headers, page 295Handling Error Responses for Outbound Calls, page 298VAR Metrics, page 300

Audio and Video File FormatsThis section provides information about the supported file formats for playing and recording audio and video media:• Audio-Only Formats—Play• Video-Only Formats—Play (see page 291)• Combined Audio and Video Formats—Play (see page 291)• Audio-Only Formats—Record (see page 292)• Video-Only Formats—Record (see page 293)• Combined Audio and Video Formats—Record (see page 294)

Audio-Only Formats—PlayTable 65 lists the supported audio-only file formats for playing prompts.


Appendix C: Media Control Platform Reference Information Audio and Video File Formats

Table 65: Supported Audio File Formats—Play

Expected File Extension

MIME-type File Format Sample Size Encoding

.vox audio/x-voxaudio/vox

Raw audio 8-bit mono G.711 ulaw, G.711 alaw (depends on platform configuration)

.au audio/basic* Audio with .au header

8-bit mono G.711 ulaw, G.711 alaw, PCM, ADPCM (depends on file header information)

.ulaw audio/basic* Raw audio 8-bit mono G.711 ulaw

.alaw audio/x-alaw-basic

Raw audio 8-bit mono G.711 alaw

.g729 audio/g729 Raw audio G.729

.pcm audio/pcmaudio/x-pcm

Raw audio 8-bit unsigned mono

Linear PCM

.adpcm24 audio/x-g726-24

Raw audio 24 kb/sec ADPCM (G.726)

.adpcm audio/x-g726audio/x-adpcmaudio/adpcmaudio/x-adpcm8


.adpcm40 audio/x-g726-40


.pcm8 audio/L8audio/pcm8audio/x-pcm8


Linear PCM

.pcm16 audio/L16 Raw audio 16-bit signed mono

Linear PCM

Note: The sample rate is always 8000 Hz. If a non-8000 Hz audio file is detected, a warning message will be issued, and the prompt will be played as if the sampling rate is 8000 Hz. A configurable Media Control Platform parameter, mpc.mediamgr.strictsamplingrate, enables you to prevent non-8000 Hz audio files from being played.

User’s Guide 291


Video-Only Formats—PlayTable 66 lists the supported video-only file formats for playing prompts.

Combined Audio and Video Formats—PlayTable 67 lists the supported audio/video file formats for playing prompts.

.wav audio/wavaudio/x-wav

Audio with .wav header

G.711 ulaw, G.711 alaw, PCM, ADPCM (depends on file header information)

.nist audio/wavaudio/x-wav

Audio with NIST header

8-bit mono G.711 ulaw, G.711 alaw (depends on file header information)

.gsm audio/x-gsm Raw audio gsm 6.10

.amr audio/amr Raw audio AMR

.3gp audio/3gpp Audio stored in 3GP container

AMR

* The Media Control Platform examines the audio data to determine whether an audio/basic file is actually of .au format or .ulaw format.

Table 65: Supported Audio File Formats—Play (Continued)


MIME-type File Format Sample Size Encoding

Note: The sample rate is always 8000 Hz. If a non-8000 Hz audio file is detected, a warning message will be issued, and the prompt will be played as if the sampling rate is 8000 Hz. A configurable Media Control Platform parameter, mpc.mediamgr.strictsamplingrate, enables you to prevent non-8000 Hz audio files from being played.

Table 66: Supported Video File Formats—Play


MIME-type Sample Rate File Format Encoding

.263 video/h263video/x-h263

30 fps (recommended)

Raw video h263

.263 video/h263-1998

30 fps (recommended)

Raw video h263-1998



Audio-Only Formats—RecordTable 68 lists the supported audio-only file formats for recording.

Table 67: Supported Audio/Video File Formats—Play


MIME-type Sample Rate File Format Encoding

.avi video/avivideo/x-avi

• Audio: 8000 Hz• Video: 30 fps

(recommended)

Audio/video stored in AVI container

• Audio: G.711 ulaw, G.711 alaw, PCM, ADPCM (depends on file header information)

• Video: h263, h263-1998 (depends on file header information)

.3gp video/3gpp • Audio: 8000 Hz• Video: 30 fps

(recommended)

Audio/video stored in 3GP container

• Audio: AMR• Video: h263, h263-1998

(depends on file header information)

Table 68: Supported Audio File Formats—Record

MIME-type Recorded File Format

Sample Size Encoding File Extension

audio/x-voxaudio/vox

Raw audio 8-bit mono G.711 ulaw, G.711 alaw (depends on platform configuration)

.vox

audio/basic Raw audio 8-bit mono G.711 ulaw .ulaw

audio/x-alaw-basic Raw audio 8-bit mono G.711 alaw .alaw

audio/g729 Raw audio G.729 .g729

audio/pcmaudio/x-pcm


Linear PCM .pcm

audio/x-g726-24 Raw audio 24 kb/sec ADPCM (G.726) .adpcm24

Notes: • Only the 8000 Hz audio sampling rate is supported.• Genesys Voice Platform (GVP) 8.0 does not support .au and .nist file recording.

User’s Guide 293


Video-Only Formats—RecordTable 69 lists the supported video-only file formats for recording.

audio/x-g726audio/x-adpcmaudio/adpcmaudio/x-adpcm8

Raw audio 32 kb/sec ADPCM (G.726) .adpcm

audio/x-g726-40 Raw audio 40 kb/sec ADPCM (G.726) .adpcm40

audio/L8 Raw audio 8-bit unsigned mono

Linear PCM .pcm8

audio/L16 Raw audio 16-bit signed mono

Linear PCM .pcm16

audio/x-wav;codec=<audio_codec>;rate=<g726_encoding_rate>audio/wav;codec=<audio_codec>;rate=<g726_encoding_rate>

Audio with .wav header

audio_codec: ulaw, alaw, pcm, pcm16, g726, gsm. Default: ulaw or alaw (depends on platform configuration).g726_encoding_rate: 16 kb, 24 kb, 32 kb, or 40 kb. Default: 32 kb.

.wav

audio/x-gsm Raw audio gsm 6.10 .gsm

audio/amr Raw audio AMR .amr

audio/3gpp Audio stored in 3GP container

AMR .3gp

Table 68: Supported Audio File Formats—Record (Continued)


Sample Size Encoding File Extension

Notes: • Only the 8000 Hz audio sampling rate is supported.• Genesys Voice Platform (GVP) 8.0 does not support .au and .nist file recording.


Appendix C: Media Control Platform Reference Information Dynamic Media Control Platform Parameters

Combined Audio and Video Formats—RecordTable 70 lists the supported audio/video file formats for recording.

Dynamic Media Control Platform Parameters

This section lists the configuration and service parameters whose values can be set dynamically for a call session.The dynamic value is obtained from the gvp.config.<parameter name>=<parameter value> parameter in the Request-URI of the establishing SIP INVITE, or from the Request-URI of a REFER that triggers an outbound call.

Table 69: Supported Video File Formats—Record

MIME-type Recorded File Format Encoding File Extension

video/h263 Raw video h263 .263

video/h263-1998 Raw video h263-1998 .263

Table 70: Supported Audio/Video File Formats—Record


Encoding File Extension

video/avi;codec=<audio_codec>;rate=<g726_encoding_rate>;videocodec=<video_codec>video/x-avi;codec=<audio_codec>;rate=<g726_encoding_rate>;videocodec=<video_codec>

Audio/video stored in AVI container

audio_codec: ulaw, alaw, pcm16, pcm8, gsm, g726, none. Default: ulaw or alaw (depends on platform configuration).video_codec: h263, h263-1998. Default: h263g726_encoding_rate: 16 kb, 24 kb, 32 kb, or 40 kb. Default: 32 kb.

.avi

video/3gpp;codec=<audio_codec>;videocodec=<video_codec>

Audio/video stored in 3GP container

audio_codec: amr, none. Default: amrvideo_codec: h263, h263-1998.Default: h263

.3gp

Note: Only the 8000 Hz audio sampling rate is supported.

User’s Guide 295

Appendix C: Media Control Platform Reference Information SIP Headers

The following configuration options can be set dynamically:

SIP HeadersTable 71 lists the SIP headers that the Media Control Platform recognizes and uses. You can use values from many of these headers to send and receive data to and from the VoiceXML or CCXML application in SIP INFO messages.Do not use the header names in Table 71 for any custom headers, or they will be ignored.

asr Section mpc Sectionload_once_per_call*

* The value can also be overridden by the value of the gvp.policy.mcp-asr-usage-mode parameter of the IVR Profile, which the Resource Manager passes to the Media Control Platform as a Request-URI parameter.

codec

delay_for_dtmf codecpref

log_metrics_to_asr fcr.defaultdtmfhandling

transmitmultiplecodec

sessmgr Section appendrejcodec

maxincalltime rtp.dtmf.receive

ECS_Fallback rtp.dtmf.send

join_fallback

record.start.beep.filename sip Sectioninbandxferprefix warningheaders

inbandxfertimeout sendalert

alert_before_fetch sendrecvevents

mediaswitch_on_alert

acceptcalltimeout

Table 71: SIP Headers Known to GVP

SIP Header Name

Standard/Specification (Section)

Description

Accept RFC 3261 (20.1) When responding to a SIP OPTIONS request, lists all the content types accepted by the component.

Allow RFC 3261 (20.5) When responding to SIP OPTIONS request, lists all the methods supported by the component.

Call-ID RFC 3261 (20.8) Standard support.



Contact RFC 3261 (20.10) Forms the remote request URI in a dialog.

Content-Length RFC 3261 (20.14) Standard support.

Content-Type RFC 3261 (20.15) Supported content types:• application/dtmf-relay

• application/sdp

• application/text

• application/www-form-urlencoded

• message/sipfrag;version=2.0

• telephone/event

CSeq RFC 3261 (20.16) Standard support.

Diversion draft-levy-sip-diversion (08) Exposed to the application as a read-only redirection variable if History-Info header is not available.

Event RFC 33515 Supported event package:• refer

From RFC 3261 (20.20) Contains the calling party information (ANI). Maps to the VoiceXML session variable session.connection.remote.uri.

History-Info RFC 4244 The list of header values that are exposed at the application layer as the redirection variable. Maps to the VoiceXML session variable session.connection.redirect.

• Original Called Number (OCN) is treated as the first entry in the History-Info header.

• Redirection Reason is treated as a list of all reasons in the History-Info header values.

Min-Expires RFC 4028 (5) Minimum session timer.

Max-Forwards RFC 3261 (20.22) Standard support.

P-Asserted-Identity

RFC 3325 Provides the calling party information (ANI) if the From header is anonymous.If this header exists, its value overrides the From header as the ANI.

Table 71: SIP Headers Known to GVP (Continued)

SIP Header Name


Description

User’s Guide 297


Privacy RFC 3323 Sets the Presentation Indicator of the VoiceXML session variable session.connection.redirect.

Reason RFC 3326 If the Reason header is in the BYE message, the reason text will be available as a read-only variable in the application.

Record-Route RFC 3261 (20.30, 16.12.1) Specifies the routeset when sending requests within the dialog.

Refer-To RFC 3515 (2.1) Sets the destination of the transfer request.

Replaces RFC 3891 Sets the dialog to replace for whisper transfer.

Require RFC 3261 (20.32) Supported option tags:• 100rel (PRACK not supported)• timer

If the Media Control Platform receives tags that it does not understand, it rejects the request with 420 Bad Extension.

Route RFC 3261 (20.34) Sets the next hop address when sending a request. The value can be set by the application or by configuration.If the INVITE contains Record-Route headers, Record-Route values override the configured routeset for all requests within the dialog.

RSeq RFC 3262 (7.1) Sent by the User Agent Server (UAS) on a reliable response.

Rack RFC 3262 (7.2) Sent by the User Agent Client (UAC) to acknowledge (ACK) a reliable response.

Session-Expires RFC 4028 (4) Sets the session expiry time and the refresher role.

Subscription-State

RFC 3515 Supported by the REFER method only.

Supported RFC 3261 (20.37) Supported option tags:• 100rel (PRACK not supported)• timer


SIP Header Name


Description


Appendix C: Media Control Platform Reference Information Handling Error Responses for Outbound Calls

Handling Error Responses for Outbound Calls

Table 72 summarizes how the Media Control Platform interprets SIP error responses that it receives in response to outgoing INVITE requests.

To RFC 3261 (20.39) Contains the called party information (DNIS). Maps to the VoiceXML session variable session.connection.local.uri.

Unsupported RFC 3261 (20.40) Contains the list of option tags not supported by the User Agent (UA) when rejecting a call.

Via RFC 3261 (20.42) Standard support.

Warning RFC 3261 (20.43) Returned by a UAS when it failed to negotiate a media session or the request contained a malformed NETANN request. The following warning codes are used in the following situations:• 300 - incompatible network protocol

• 301 - incompatible network address

• 302 - incompatible transport protocol

• 303 - incompatible bandwidth

• 304 - unsupported media type

• 305 - unsupported media format

• 306 - unknown attribute not supported

• 307 - unknown parameter was presented

• 399 - malformed request URI (malformed NETANN request)

X-Genesys-CallUUID

Genesys UUID, generated by SIP Server (or T-Server).

X-Genesys-GVP-Session-ID

GVP Session Identifier, generated by the Resource Manager (for new inbound sessions) or the Media Control Platform or the Call Control Platform (for new outbound sessions).

X-Genesys-RM-Application-dbid

The DBID of the IVR Profile (in other words, VoiceXML or CCXML application).


SIP Header Name


Description

User’s Guide 299

Appendix C: Media Control Platform Reference Information Handling Error Responses for Outbound Calls

For information about SIP response codes that the Media Control Platform generates, see Appendix B, “SIP Response Codes,” on page 279.

Table 72: Error Response Handling—Outbound Calls

SIP Response Call End Reason (for Metrics)

Disconnect Reason (to Determine Call/Transfer Result)

Action in VoiceXML Application

Code Phrase

301 Moved Permanently

baddest CM_DISCREASON_BADDEST

error.connection.baddestination event during <transfer>

404 Not Found

410 Gone

484 Address Incomplete

502 Bad Gateway

401 Unauthorized noautho CM_DISCREASON_OUT_NOAUTH

error.connection.noroute event during <transfer>A <transfer> form value of unknown is assigned.

402 Payment Required

403 Forbidden

407 Proxy Authentication Required

408 Request Timeout

noanswer CM_DISCREASON_OUT_NOANSWER

noanswer in the <transfer> result

480 Temporarily Unavailable

busy CM_DISCREASON_OUT_USERBUSY

busy in the <transfer> result

486 Busy Here

405 Method Not Allowed

unsupported CM_DISCREASON_UNSUPPORTED

error.unsupported.transfer.blind/consultation/bridge event during <transfer>

488 Not Acceptable Here

501 Not Implemented


Appendix C: Media Control Platform Reference Information VAR Metrics

VAR MetricsTable 73 summarizes the metrics that the Media Control Platform generates when the Next Generation Interpreter (NGI) executes a VAR-specific <log> tag. The metrics include the PCDATA specified in the <log> element.Table 73 includes information about the valid syntax and values for the VAR-specific <log> tag. If the format of the PCDATA for the element does not conform to the valid syntax, the VAR metric will not be logged.For more information about using the VAR <log> tag labels (or extensions) in VoiceXML applications, see the Genesys Voice Platform 8.0 VoiceXML 2.1 Help.

Formatting Note

Contrary to type conventions in the remainder of this guide, italic text in the <log> tag syntax indicates placeholders for user-specified values. The angle brackets are a required part of the VoiceXML syntax.

503 Service Unavailable

resourcelimit CM_DISCREASON_OUT_NORESRC

error.connection.noresource event during <transfer>

504 Gateway Timeout

busy CM_DISCREASON_OUT_NWBUSY

network_busy in the <transfer> result

No response

All other errors error CM_DISCREASON_GENERROR

error.connection.noroute event during <transfer>A <transfer> form value of unknown is assigned.

Table 72: Error Response Handling—Outbound Calls (Continued)

SIP Response Call End Reason (for Metrics)

Disconnect Reason (to Determine Call/Transfer Result)

Action in VoiceXML Application

Code Phrase

User’s Guide 301


Table 73: VAR <log> Tags and Metrics

Metric <log> Tag Label Syntax and Valid Values

call_result <log label="com.genesyslab.var.CallResult">result[|reason]</log>

where:• result is SUCCESS|FAILED|UNKNOWN. (The default is UNKNOWN.)• reason is an optional string of up to 256 characters that provides a textual reason

for the call result.Notes• result and reason values are not case-sensitive.• If the developer specifies a call result other than SUCCESS or FAILED, UNKNOWN is

assumed.• Preceding and trailing whitespace in the result is ignored.• reason content beyond 256 characters will be truncated.

call_notes <log label="label=com.genesyslab.var.CallNotes">notes</log>

where notes are up to 4 KB (4096 bytes) of free-form notes associated with the call.Notes• notes cannot be empty.• Content beyond 4 KB will be truncated.

ivr_action_start <log label="com.genesyslab.var.ActionStart">actionID[|parentID=PID]</log>

where:• actionID is the ID of the VoiceXML application action being started.• PID is the ID of the parent action, if this action is nested inside some other active

action.Notes• actionID and PID are any valid UTF8 string, to a maximum of 64 characters, that

does not contain spaces or pipes.• Action IDs are case-sensitive.• Whitespace is ignored.• An active action is an action that has started and has not yet ended. If a specified

PID is not the ID of an active action, the reporting infrastructure will ignore the ivr_action_start metric.



ivr_action_end <log label="com.genesyslab.var.ActionEnd">actionID[|result[|reason]]</log>

where:• actionID is the ID of the VoiceXML application action being ended.• result is one of SUCCESS|FAILED|UNKNOWN, indicating the result of the action. The

default is UNKNOWN.• reason is an optional string of up to 256 characters that provides a textual reason

for the action result.If ActionEnd is not explicitly specified, the Reporting Server implicitly ends actions under the following circumstances:• A sibling action (in other words, an action with the same parent) is started.• The call ends.If the Reporting Server implicitly ends an action, the value of result is UNKNOWN, and the value of reason is NULL.Notes• actionID is any valid UTF8 string, to a maximum of 64 characters, that does not

contain spaces or pipes.• actionID is case-sensitive.• result and reason values are not case-sensitive.• Whitespace in the metric is ignored.• If the specified actionID is not the ID of an active action, the reporting

infrastructure will ignore the ivr_action_end metric.• If the developer specifies an action result other than SUCCESS, FAILED, or UNKNOWN,

the reporting infrastructure will ignore the ivr_action_end metric.

Table 73: VAR <log> Tags and Metrics (Continued)


User’s Guide 303


ivr_action_notes <log label="com.genesyslab.var.ActionNotes">actionID|notes</log>

where:• actionID is the ID of the VoiceXML application action.• notes are up to 4 KB (4096 bytes) of free-form notes associated with the action.Notes• actionID is any valid UTF8 string, to a maximum of 64 characters, that does not

contain spaces or pipes. Preceding and trailing whitespace is ignored.• notes cannot be empty.• notes content beyond 4 KB will be truncated.• VoiceXML action notes (ivr_action_notes) may be logged during the specified

action or after it has ended.

custom_var <log label="com.genesyslab.var.CustomVar">name|value</log>

where:• name is the name of the custom variable.• value is the value of the custom variable.Notes• name is any valid UTF8 string, to a maximum of 64 characters, that does not

contain spaces or pipes. Preceding and trailing whitespace is ignored.• value is any valid UTF8 string, to a maximum of 256 characters. Whitespace is

significant.• Custom variables may be specified at any point in a VoiceXML application.• The reporting infrastructure will allow a maximum of eight (8) custom variables to

be specified for a given call. Any variables logged beyond the maximum will be ignored.

Table 73: VAR <log> Tags and Metrics (Continued)


User’s Guide 304

Appendix

D Default Device ProfilesThis appendix summarizes the settings for the default device profiles that are provisioned for the Call Control Platform. These device profiles are defined in the <Call Control Platform Installation Directory>\config\ccpccxml_provision.dat file.Table 74 summarizes the settings, by profile.The format of each entry in the profile definition file is:<entry id="Entry ID" type="401" name="CCXML Device Profile">PrecedenceProfile NameSIP DeviceNumber of propertiesPropertyA ValueAPropertyB ValueB...User-Agent User-Agent</entry>

Where:• The angle brackets are a necessary part of the syntax.• Italic text indicates placeholders for items that are listed in Table 74.• SIP Device is the Device Profile Class Name.• User-Agent is the SIP Header Name.You may optionally add a short description in each property line. For descriptions of the properties, see Table 33 on page 189.

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P104


Appendix D: Default Device Profiles

User’s Guide 307

Appendix

E Specifications and StandardsThis appendix describes the specifications and standards that GVP supports.It contains the following sections:

Specifications, page 307Burke Draft Support, page 309

SpecificationsThe following specifications are published and maintained by the W3C Voice Browser Working Group:• VoiceXML Specification—W3C Voice Extensible Markup Language

(VoiceXML) 2.1, W3C Recommendation 19 June 2007 and W3C Voice Extensible Markup Language (VoiceXML) 2.0, W3C Recommendation 16 March 2004

• Speech Synthesis Markup Language Specification—W3C Speech Synthesis Markup Language (SSML) Version 1.0, Recommendation 7 September 2004

• Speech Recognition Grammar Specification—Speech Recognition Grammar Specification Version 1.0, W3C Recommendation 16 March 2004

• CCXML Specification—W3C Voice Browser Call Control: CCXML Version 1.0, W3C Working Draft 29 June 2005


Appendix E: Specifications and Standards Related Standards

Related StandardsGVP is based on open standards. As a result, the platform provides complete or subset support for many Requests for Comments (RFCs) that are published and maintained by the Internet Engineering Task Force (IETF). For more information, see http://www.ietf.org.The IETF standards that GVP supports include:• RFC 1738 Uniform Resource Locators• RFC 1808 Relative Uniform Resource Locators• RFC 1867 Form-based File Upload in HTML• RFC 2109 HTTP State Management Mechanism• RFC 2190 RTP Payload Format for H.263 Video Streams• RFC 2388 Returning Values from Forms: multipart/form-data• RFC 2326 Real Time Streaming Protocol (RTSP)• RFC 2396 Uniform Resource Identifiers (URI): Generic Syntax• RFC 2429 RTP Payload Format for the 1998 Version of ITU-T Rec. H.263

Video (H263+)• RFC 2616 Hypertext Transfer Protocol – HTTP/1.1 (subset)• RFC 2806 URLs for Telephone Calls• RFC 2833 RTP Payload for DTMF Digits, Telephony Tones and

Telephony Signals• RFC 2964 Use of HTTP State Management• RFC 2965 HTTP State Management Mechanism• RFC 2976 The SIP INFO Method• RFC 3261 SIP: Session Initiation Protocol• RFC 3264 An Offer/Answer Model with the Session Description Protocol

(SDP)• RFC 3267 Real Time Transport Protocol (RTP) Payload Format and File

Storage Format for the Adaptive Multi-Rate (AMR) and Adaptive Multi-Rate Wideband (AMR-WB) Audio Codecs

• RFC 3515 The SIP REFER Method• RFC 3550 RTP: A Transport Protocol for Real-Time Applications• RFC 3986 Uniform Resource Identifier (URI): Generic Syntax• RFC 4240 Basic Network Media Services with SIP—GVP provides

support for conference service and dialogs• RFC 4568 Session Description Protocol (SDP) Security Descriptions for

Media Streams

User’s Guide 309

Appendix E: Specifications and Standards Burke Draft Support

• Most extensions from proposed IETF draft SIP Interface to VoiceXML Media Services (http://tools.ietf.org/id/draft-burke-vxml-02.txt)—see “Burke Draft Support”

• Some extensions from proposed IETF draft Basic Network Media Services with SIP (http://www.ietf.org/rfc/rfc4240.txt)

The platform also includes complete support for many network-related and other protocols (for example, TCP/IP, SNMP). Contact Genesys for additional details.

Burke Draft SupportThis section describes various aspects of the Burke Draft (http://www.ietf.org/internet-drafts/draft-ietf-mediactrl-vxml-01.txt), and the features supported by GVP 8.0.The Burke Draft describes the SIP interface to VoiceXML media services. The following Burke Draft sections are covered:• 2—“VoiceXML Session Establishment and Termination”• 3 —“Media Support” on page 311• 4 — “Returning Data to the Application Server” on page 312• 5 — “Outbound Calling” on page 313• 6 — “Call Transfer” on page 313Table 75 lists and describes the Burke Draft requirements and which ones GVP 8.0 supports.

Table 75: Burke Draft Support Requirements, by Section

Section Requirement Current Support

VoiceXML Session Establishment and Termination

2.1 Support for the following service identification parameters:• voicexml

• maxage [RFC2161]• maxstale [RFC2616]• method

• postbody

• ccxml [RFC4627]• aai [RFC4627]

Supported

Support for incorrectly formed requests with the 4xx class response.

Not Supported



2.1 (continued)

Support for repeated init-parameters rejected with the 400 Bad Request response.

Supported

Support for parameter URL-encoding. Supported

2.2 Support for 100 Trying, followed by a 200 OK response upon receipt of a INVITE when a document has been fetched. After the ACK is received, the application executes.

Supported

Support for optimization before sending the 200 OK response. Not Supported

Support for the inability to accept INVITE requests, and to respond as defined by [RFC3261] with the exception of the following error conditions:• If the request does not conform to the specification, return a 400 Bad Request response.

• If the request does not include a voicexml parameter, and the default page is not configured, return a 400 Bad Request and a 399 Warning header.

Not Supported

Support for returning a 500 Server Internal Error with a 399 Warning header if the document cannot be fetched or parsed.

Supported

Support for large transport appropriate messages (such as TCP) when an INVITE request exceeds the MTU of the underlying network.

Supported

2.3 Support for not exiting VoiceXML application until a re-INVITE with port information is sent if starting a Dialog-INVITE without media; a 200 OK with offered media; an ACK with media, but the media ports are set to zero (0); or an INVITE with SDP without media lines followed by a regular INVITE, 200 OK, or ACK flow.

Supported

Support for a re-INVITE that disables media stream without affecting the executing VoiceXML application once the application is running.

Supported

2.4 Support for the following session variables:• session.connection.local.uri

• session.conection.remote.uri

• session.connection.redirect

• session.connection.protocol.sip.headers

• session.connection.aai

• session.connection.ccxml

Supported

Table 75: Burke Draft Support Requirements, by Section (Continued)


User’s Guide 311


2.4 (continued)

Support for evaluating the following session variables:• session.connection.protocol.name to sip.• session.connection.protocol.version to 2.0

Supported

Support for using the session.connection.protocol.sip.requesturi variable as an associative array formed from the URI parameters.

Supported

Support for using the session.connection.protocol.sip.media variable as an array where each array element is an object with the following properties:• type

• direction

• format

Note: This parameter will be updated as the media values involved in the session change.

Supported

2.5 Support for sending a 200 OK in response to a BYE, and then generating a connection.disconnect.hangup event.

Supported

Support for providing the value of the Reason header verbatim through the _message variable if a Reason header [RFC3326] is present in the BYE Request.Note: Set sip.in.bye.headers to Reason.

Supported

Support for terminating a session with a BYE Request when the VoiceXML application encounters a <disconnect> or <exit>, the VoiceXML application completes, or the VoiceXML application has unhandled errors.

Supported

Media Support

3.1 Support for the Offer/Answer mechanism of [RFC3960]. Supported

3.2 Support for early media streams as described in [RFC3960]. Supported

3.3 Support for the using a re-INVITE to modify a media session. Supported





3.4 Support for the following Codecs:• G.711 mu-law and A-law using the RTP payload type 0 and 8—

set mpc.codec to pcmu pcma• H.263 Baseline—set mpc.codec to h263• AMR-NB audio—set mpc.codec to amr• Other codecs and payload formats—set mpc.codec as specified

for the various codecs.

Supported

Support for the following Codecs:• MPEG-4 video• MPEG-4 AAC audio

Not Supported

3.5 Support for DTMF events [RFC4733]. Supported

Returning Data to the Application Server

4.1 Support for returning data to the application server with a HTTP Post using the <submit>, <subdialog>, or <data> elements.

Supported

4.2 Support for returning data to the application server with the SIP expression or the namelist attribute on the <exit> element, or the namelist attribute on the <disconnect> element.

Supported

Support for encoding the expr or the namelist data in the BYE request body when encountering the <exit> or <disconnect> element.

Supported

Support for including the expr or the namelist data in the 200 OK response to a BYE request.

Not Supported

Support for sending a 100 Trying response to a BYE request [RFC4320].

Not Supported

Support for using the _reason reserved name to differentiate between a BYE resulting from a <disconnect> element and a BYE resulting from an <exit> element.For example, _reason = exit.

Supported

Support for using the _exit reserved name if the expr attribute is specified on the <exit> element instead of the namelist attribute.For example, _exit=<value>

Supported



User’s Guide 313


Outbound Calling

5.0 Support for triggering outbound calls using third party call control [RFC3725].

Supported

Call Transfer

6.1 Support for blind transfers with a REFER message on the original SIP dialog [RFC3515]. Note: Set sip.defaultblindxfer to REFER.

Supported

Support for terminating the session with a BYE message and generating the connection.disconnect.transfer event if a REFER request is accepted with a 2xx response.

Supported

Support for the following form item variables and events depending on the SIP response if the REFER is accepted with a non-2xx response:• 404 Not Found = error.connection.baddestination• 405 Method Not Allowed = error.unsupported.transfer.blind• 503 Service Unavailable = error.connection.noresource• (No response) = network_busy• (Other 3xx/4xx/5xx/6xx) = unknown

Supported

Support for appending the aai or aaiexpr attribute to the Refer_To URI as a parameter named aai.

Supported

Support for URL-encoding reserved characters as required for SIP/SIPS URIs [RFC3261].

Supported

6.2 Support for ejecting the callee if the bridged transfer is terminated. Supported

Support for appending the aai or aaiexpr attribute to the Refer_To URI in the INVITE as a parameter named aai.

Supported

Support for URL-encoding reserved characters as required for SIP/SIPS URIs [RFC3261].

Supported

Supporting for playing early media from the callee to the caller if the transferaudio attribute is omitted.

Supported

Support for setting the <transfer>’s form attribute to noanswer after issuing a CANCEL when connectiontimeout expires.

Supported





6.2 (continued)

Support for the following form item variables and events depending on the SIP response if the INVITE is accepted with a non-2xx response:• 404 Not Found = error.connection.baddestination• 405 Method Not Allowed = error.unsupported.transfer.blind• 408 Request Timeout = noanswer• 486 Busy Here = busy• 503 Service Unavailable = error.connection.noresource• (No response) = network_busy• (Other 3xx/4xx/5xx/6xx) = unknown

Supported

Support for listening for speech or DTMF hotword results in a near end disconnect.

Supported

Support for issuing a BYE if the call duration exceeds the maximum duration specified in the maxtime attribute.

Supported

6.3 Support for the following form item variables and events depending on the SIP response if the INVITE is accepted with a non-2xx response:• 404 Not Found = error.connection.baddestination• 405 Method Not Allowed = error.unsupported.transfer.consultation

• 408 Request Timeout = noanswer• 486 Busy Here = busy• 503 Service Unavailable = error.connection.noresource• (No response) = network_busy• (Other 3xx/4xx/5xx/6xx) = unknown

Supported

Support for generating the connection.disconnect.transfer event when receiving a 200 OK to a NOTIFY request.

Supported

Support for setting the VoiceXML input item variable to unknown with the non-2xx response to the NOTIFY request.

Not Supported



User’s Guide 315

Appendix

F Caching Reference InformationThis appendix provides information about the Fetching Module and Squid caching algorithms, as well as information about the Squid access log.It contains the following sections:

Caching Algorithms, page 315Squid Access Logs, page 317

Caching AlgorithmsThis section provides the algorithms that the Fetching Module and Squid use to determine when they will use cached documents:• Fetching Module Caching Algorithm• Squid Caching Algorithm (see page 316)• Squid Expiry Time Algorithm (see page 316)

Fetching Module Caching AlgorithmWhen the VoiceXML interpreter requests the Fetching Module to perform a fetch to URI, the Fetching Module uses the following algorithm to determine whether it will use the cached version within its own memory:if ( URI contains one of the items in the iproxy.no_cache_url_substr list )

re-fetch the item from the squid proxyelse

if (the URI matches exactly [including all parameters] with a URI in Fetching Module cache)

if (the previous was a fetch error)


Appendix F: Caching Reference Information Caching Algorithms

if (the previous result was fetched within iproxy.cache_error_max_age seconds)

return result from cache;end ifelse if (the previous was a successful fetch)if (the previous result was fetched within iproxy.cache_max_age

seconds)return result from cache;end ifend ifend ifre-fetch the item from the squid proxyend if

Squid Caching AlgorithmThe following algorithm summarizes the caching policy rules, and represents the Squid caching proxy behavior when a resource is requested by the VoiceXML interpreter.If the resource is not present in the cache, fetch it from the server using get.

If the resource is in the cache,If a maxage value is provided,If age of the cached resource <= maxage,If the resource has expired,Perform maxstale check.Otherwise, use the cached copy.Otherwise, fetch it from the server using get.

Otherwise,If the resource has expired,Perform maxstale check.Otherwise, use the cached copy.

The algorithm for maxstale check is:If maxstale is provided,

If cached copy has exceeded its expiration time by no more than maxstale seconds,

then use the cached copy.Otherwise, fetch it from the server using get.

Otherwise, fetch it from the server using get.

Squid Expiry Time AlgorithmDetermining the freshness or staleness of an object is a multi-step process:1. Squid calculates the following values:

AGE—How much the object has aged since it was retrieved.(AGE = NOW - OBJECT_DATE)

User’s Guide 317

Appendix F: Caching Reference Information Squid Access Logs

LM_AGE—How old the object was when it was retrieved.(LM_AGE = OBJECT_DATE - LAST_MODIFIED_TIME)LM_FACTOR—The ratio of AGE to LM_AGE.(LM_FACTOR = AGE / LM_AGE)CLIENT_MAX_AGE—(Optional) The maximum object age the client will accept, as taken from the Cache-Control request header.EXPIRES—(Optional) The expiry time from the server reply headers.

2. Squid compares the calculated values against the parameters of the refresh_pattern rules (see “Refresh-Pattern Rules” on page 201). Squid checks the URL regular expressions in the order in which the refresh-pattern rules are listed, until a match is found. Squid uses the first entry that matches. If no match is found, the following default values are used:

MAX_AGE = 4320 minutesMIN_AGE = 0PERCENT = 20%

3. When a match is found, Squid applies the following algorithm to determine if the object is fresh or stale:if (CLIENT_MAX_AGE)if (AGE > CLIENT_MAX_AGE)return STALE

if (AGE <= MIN_AGE)return FRESH

if (EXPIRES) {if (EXPIRES <= NOW)return STALE

elsereturn FRESH

}if (AGE > MAX_AGE)return STALE

if (LM_FACTOR < PERCENT)return FRESH

return STALE

Note: The MAX-AGE in a client request takes the highest precedence. Genesys recommends that you usually set the MIN_AGE value to zero, because it has higher precedence than the Expires: value from the server. However, use the MIN_AGE value if you wish to override the Expires: headers.

Squid Access LogsThe Squid access.log is stored in the following location:C:\squid\var\logs\



Table 76 describes the fields in each access.log entry.

Table 77 describes the log tags for the following types of requests:• TCP requests, on the HTTP port• UDP requests, on the ICP port

Table 76: Squid Access Log Fields

Field Description

Timestamp The time when the client socket is closed. The format is “Unix time” (seconds since Jan 1, 1970), with millisecond resolution. Use the following command to modify this to visible format:cat access.log | perl -nwe ‘s/^(\d+)/localtime($1)/e; print’

Elapsed Time The elapsed time of the request, in milliseconds. This is time between the accept() and close() of the client socket.

Client Address The IP address of the connecting client, or the fully qualified domain name (FQDN) if the log_fqdn option is enabled in the configuration file. This parameter is normally turned off for performance reasons.

Log Tag/HTTP Code The Log Tag describes how the request was treated locally (hit, miss, and so on). The HTTP code is the reply code taken from the first line of the HTTP reply header. Non-HTTP requests may have zero reply codes.For descriptions of the Log Tags, see Table 77 on page 318.

Size The number of bytes written to the client.

Request Method The HTTP request method, or ICP_QUERY for ICP requests.

URL The requested URL.

Ident If ident_lookup is on, this field may contain the username associated with the client connection as derived from the ident service. This lookup is typically turned off for performance reasons.

Hierarchy Data/ Hostname

A description of how and where the requested object was fetched.

Content Type The Content-type field from the HTTP reply.

Table 77: Log Tags in the Squid Access Logs

Access Log Tag Description

TCP_ Requests

TCP_HIT A valid copy of the requested object was in the cache.

User’s Guide 319


TCP_MISS The requested object was not in the cache.

TCP_REFRESH_HIT The object was in the cache, but STALE. An If-Modified-Since request was made, and a 304 Not Modified reply was received.

TCP_REF_FAIL_HIT The object was in the cache, but STALE. The request to validate the object failed, so the old (stale) object was returned.

TCP_REFRESH_MISS The object was in the cache, but STALE. An If-Modified-Since request was made and the reply contained new content.

TCP_CLIENT_REFRESH The client issued a request with the no-cache pragma.

TCP_CLIENT_REFRESH_MISS

The client issued a no-cache pragma, or some analogous cache control command along with the request. Thus, the cache has to refetch the object from origin server. In short, the browser forced the proxy to check for a new version (usually caused by users clicking Reload, but can also be triggered in some browsers by selecting a bookmark).

TCP_IMS_HIT The client issued an If-Modified-Since request and the object was in the cache and still fresh. TCP_HIT and TCP_IMS_HIT are hits, the only difference is that in the TCP_IMS_HIT case the browser already had an up-to-date version so there was no need to send the Squid cached copy to the requestor.

TCP_IMS_MISS The client issued an If-Modified-Since request for a stale object.

TCP_NEGATIVE_HIT A previously failed request is satisfied from the cache, as the proxy believes it still be a problem.

TCP_SWAPFAIL The object was believed to be in the cache, but could not be accessed.

TCP_DENIED Access was denied for this request.

UDP_ Requests

UDP_HIT A valid copy of the requested object was in the cache.

UDP_HIT_OBJ Same as UDP_HIT, but the object data was small enough to be sent in the UDP reply packet. Saves the next TCP request.

UDP_MISS The requested object was not in the cache.

UDP_DENIED Access was denied for this request.

Table 77: Log Tags in the Squid Access Logs (Continued)




UDP_INVALID An invalid request was received.

UDP_RELOADING The ICP request was “refused” because the cache is busy reloading its metadata.

Table 77: Log Tags in the Squid Access Logs (Continued)


User’s Guide 321

Index

Symbols<log> tags

interface . . . . . . . . . . . . . . . . . . . 63syntax . . . . . . . . . . . . . . . . . . . 300VAR . . . . . . . . . . . . . . . . . . . . . 63

<protocol>_proxy configuration option . . . . 199<service>.<param- name> configuration

option . . . . . . . . . . . . . . . . 150<service>-capability-requirement configuration

option . . . . . . . . . . . . . . . . 145<service>-forbidden-respcode configuration

option . . . . . . . . . . . . . . . . 146<service>-forbidden-set-alarm configuration

option . . . . . . . . . . . . . . . . 146<service>-usage-limit configuration option . . 146<service>-usage-limit-exceeded-respcode

configuration option . . . . . . . . . 146<service>-usage-limit-exceeded-set-alarm

configuration option . . . . . . . . . 147<service>-usage-limit-per-session

configuration option . . . . . . . . . 147

Numerics100 Trying . . . . . . . . . . . . . . . . . . 279180 Ringing . . . . . . . . . . . . . . . . . 280183 Session Progress . . . . . . . . . . 43, 280200 OK . . . . . . . . . . . . . . . . . . .41, 43202 Accepted. . . . . . . . . . . . . . . . . 2812xx SIP response code. . . . . . . . . . . . 36302 Moved Temporarily . . . . . . . . . . . 2813xx SIP response code. . . . . . . . . . . . 281400 Bad Request. . . . . . . . . . . . . . . 281403 Forbidden . . . . . . . . . . . . . . . . 282404 Not Found . . . . . . . . . . . . 30, 32, 282405 Method Not Allowed . . . . . . . . . . . 282408 Request Timeout . . . . . . . . . . 36, 283420 Bad Extension . . . . . . . . . . . . . . 283423 Interval Too Brief. . . . . . . . . . . . . 283480 Temporarily Unavailable . . . . . . . . . 283

487 Request Terminated. . . . . . . . . . . 284488 Not Acceptable Here . . . . . . . . . . 2844xx SIP response code . . . . . . . . . . . .36500 Server Internal Error . . . . . . . . . . 285503 Service Unavailable . . . . . . . . . . . 2855xx SIP response code . . . . . . . . . . . .366xx SIP response code . . . . . . . . . . . .36

Aacceptcalltimeout configuration option. . . . 120accepting DTMF digits . . . . . . . . . . . . .41access control

configuring, for reporting . . . . . . . . . 211for reporting . . . . . . . . . . . . . . . . . 69

access log files, Squid . . . . . . . . . . 60, 317accessing Genesys Administrator . . . . . . .24active call list report . . . . . . . . . . . . . 228adding resources to groups . . . . . . . . . 129AgentX subagent . . . . . . . . . . . . . . .70ALAW . . . . . . . . . . . . . . . . . . . . 166algorithms, caching . . . . . . . . . . . . . 315all configuration option. . . . . . . . . . . . 112alternatevoicexml configuration option. . . . 151amr. . . . . . . . . . . . . . . . . . . . . . .44angle brackets . . . . . . . . . . . . . . . . .15aor configuration option . . . . . . . . . . . 129Apache. . . . . . . . . . . . . . . . . . . . .95appendrejcodec configuration option . . . . 165application DBID . . . . . . . . . . . . . . . .73application templates . . . . . . . . . . . . 160application/x-www-form-urlencoded . . . . . .52application-confmaxsize configuration

option . . . . . . . . . . . . . 103, 142applications

identifiers . . . . . . . . . . . . . . . . . . 73architecture

reporting . . . . . . . . . . . . . . . . . . 62ASR

configuration options . . . . . . . . . . . 179enabling . . . . . . . . . . . . . . . . . . 159

Index


in GVP. . . . . . . . . . . . . . . . . . . . 44per utterance or per call . . . . . . . . . . 144provisioning resources . . . . . . . . . . 160usage mode . . . . . . . . . . . . . . . . 144

asr configuration section . . . . . . . . . . . 164assigning

default IVR Profile . . . . . . . . . . . . . 154device profiles . . . . . . . . . . . . . . . . 54MRCP server . . . . . . . . . . . . . . . 162resources to groups . . . . . . . . . .129, 133

attributes (<script> tag)charset. . . . . . . . . . . . . . . . . . . . 41

attributes (CCXML)hints . . . . . . . . . . . . . . . . . . . . 117

attributes (CDR)Call Control Platform . . . . . . . . . . . . 67common set . . . . . . . . . . . . . . . . . 66Media Control Platform . . . . . . . . . . . 67

attributes (XML)dest . . . . . . . . . . . . . . . . . . . . . 53encoding. . . . . . . . . . . . . . . . . . . 41maxage . . . . . . . . . . . . . . . . . 56, 57maxstale . . . . . . . . . . . . . . . . . 56, 57method . . . . . . . . . . . . . . . . . . . 47userdata . . . . . . . . . . . . . . . . . . . 42

audience, defining . . . . . . . . . . . . . . 12audio

file formats, play . . . . . . . . . . . . . . 289file formats, record. . . . . . . . . . . . . 292files for DTMF . . . . . . . . . . . . . . . . 41recording . . . . . . . . . . . . . . . . . . 43services . . . . . . . . . . . . . . . . . . . 43

audio/videofile formats, play . . . . . . . . . . . . . . 291file formats, record. . . . . . . . . . . . . 294mixed services. . . . . . . . . . . . . . . . 43

audio_format configuration option . . . . . . 164Audiocodes Gateway (device profile) . . 54, 305Audiocodes MP104 (device profile) . . . 54, 305Automatic Speech Recognition

See ASR

Bback-off, exponential . . . . . . . . . . . . . 68barge-in . . . . . . . . . . . . . . . . . . . 43batching CDRs and metrics . . . . 68, 107, 108blind transfer type . . . . . . . . . . . . .45, 47brackets . . . . . . . . . . . . . . . . . . . 15BRIDGE transfer . . . . . . . . . . . . . .46, 48bridge transfer type. . . . . . . . . . . . .45, 48bridge_server configuration option . . . . . . 187bridge_server.profile configuration option . . 187bridged (CDR call type) . . . . . . . . . . . 67bridging services . . . . . . . . . . . . . . . 52Brooktrout Snowshore (device profile) . . 54, 305

builtin grammars . . . . . . . . . . . . . . . .45BYE message . . . . . . . . . . . . . . . . 286

CCache-Control headers . . . . . . . . 56, 57, 59caching

algorithms . . . . . . . . . . . . . . . . . 315and HTTPS . . . . . . . . . . . . . . . . . 71behavior . . . . . . . . . . . . . . . . . . . 57clearing the cache . . . . . . . . . . . . . 60expiry time algorithm . . . . . . . . . . . 316Fetching Module . . . . . . . . . . . . . . 55HTTP/1.1-compliant. . . . . . . . . . . . . 56initial page. . . . . . . . . . . . . . . . 52, 58log files . . . . . . . . . . . . . . . . . . . 60managing the cache . . . . . . . . . . . . 60non-HTTP/1.1-compliant . . . . . . . . . . 56policies . . . . . . . . . . . . . . . . . . . 56purging objects . . . . . . . . . . . . . . . 60refreshing objects . . . . . . . . . . . . . . 60Refresh-Rate model . . . . . . . . . . . . 59root page . . . . . . . . . . . . . . . . . . 52Squid . . . . . . . . . . . . . . . . . . . . 55

callarrivals statistics . . . . . . . . . . . . . . 68peaks statistics . . . . . . . . . . . . . . . 68

call completion summary report . . . . . . . 247Call Control Platform

CDR attributes . . . . . . . . . . . . . . . 67CDRs . . . . . . . . . . . . . . . . . . . . 53components . . . . . . . . . . . . . . . . . 50conference configuration options . . . . . 103configuring . . . . . . . . . . . . . . . . 184core executable . . . . . . . . . . . . . . . 50customizing SIP responses . . . . . . . . 117default SIP transport . . . . . . . . . . . . 93device profiles, default . . . . . . . . . . 304functions . . . . . . . . . . . . . . . . . . 22I/O processor . . . . . . . . . . . . . . . . 50logs . . . . . . . . . . . . . . . . . . . . . 53metrics . . . . . . . . . . . . . . . . . . . 53module IDs . . . . . . . . . . . . . . . . 267outgoing connections . . . . . . . . . . . . 53provisioning device profiles . . . . . . . . 194Reporting Client. . . . . . . . . . . . . . . 63role and functioning . . . . . . . . . . . . . 51role in call flow . . . . . . . . . . . . . . . 26session timers. . . . . . . . . . . . . . . 119SNMP . . . . . . . . . . . . . . . . . . . . 70specifier IDs. . . . . . . . . . . . . . . . 267upstream metrics . . . . . . . . . . . . . . 65

call detail recordsSee CDRs

call events, defined . . . . . . . . . . . . . .65call flow, sample . . . . . . . . . . . . . . . .25

Index

User’s Guide 323

Call Progress Analysis (CPA) . . . . . . . . 43call recording . . . . . . . . . . . . . . . . . 43call types, in CDRs . . . . . . . . . . . . . . 66calllog configuration section . . . . . . . . . 162call-timeout configuration option . . . . . . . 208CallUUID . . . . . . . . . . . . . . . . . . . 72capabilities

requirements specified . . . . . . . . . . . 34service . . . . . . . . . . . . . . . . . 33, 145SIP device . . . . . . . . . . . . . . . . . . 53

capability configuration option . . . . . . . . 131categories (of reports) . . . . . . . . . . . . 69ccpccxml configuration section. . . . . 185, 186ccpccxml.default_uri configuration option . . 51ccpccxml.exe . . . . . . . . . . . . . . . . . 50ccpccxml_provision.dat . . . . . . . . . . . 54ccpccxml_provision.dat file. . . . . . . . . . 189ccxml

service selected . . . . . . . . . . . . . . . 31service, configuring . . . . . . . . . . . . 133

CCXML applicationsand IVR Profiles . . . . . . . . . . . . . . 139debugging . . . . . . . . . . . . . . . . . . 53default . . . . . . . . . . . . . . . . . . . 140hints . . . . . . . . . . . . . . . . . . . . 117modifying for TLS . . . . . . . . . . . . . . 94provisioning . . . . . . . . . . . . . . . . 139triggering . . . . . . . . . . . . . . . . . . 77

CCXML devices . . . . . . . . . . . . . . . 189CCXML hints

outboundproxy. . . . . . . . . . . . . . . . 53CCXML Interpreter

See CCXMLIccxml parameter . . . . . . . . . . . . . . . 51CCXMLI

described . . . . . . . . . . . . . . . . . . 51role and functioning . . . . . . . . . . . . . 52role in call flow. . . . . . . . . . . . . . . . 26shared memory . . . . . . . . . . . . . . . 55

ccxmli configuration section . . . . . . 185, 187ccxml-usage-limit-exceeded-respcode

configuration option . . . . . . . . . 116ccxml-usage-limit-exceeded-set-alarm

configuration option . . . . . . . . . 116cdr configuration section . . . . . . . . . . . 208CDR Service

role and functioning . . . . . . . . . . . . . 66CDRs

and metrics . . . . . . . . . . . . . . . . . 64attributes. . . . . . . . . . . . . . . . . . . 66batching . . . . . . . . . . . . . . . . 68, 108call types . . . . . . . . . . . . . . . . . . 66delivery . . . . . . . . . . . . . . . . . . . 66described . . . . . . . . . . . . . . . . . . 66generated . . . . . . . . . . . . . . . . 42, 53interface . . . . . . . . . . . . . . . . . . . 62

persistent queue path . . . . . . . . . . . 108certificate

creating . . . . . . . . . . . . . . . . . . . 96creating self-signed . . . . . . . . . . . . . 97default path . . . . . . . . . . . . . . . . . 97SSL . . . . . . . . . . . . . . . . . . . . . 95

chapter summaries . . . . . . . . . . . . . .12charset attribute . . . . . . . . . . . . . . . .41Chinese, encoding for . . . . . . . . . . . . .41Cisco Gateway (device profile) . . . . . 54, 305clearing the cache . . . . . . . . . . . . . . .60client.ConnectPerSetup configuration

option . . . . . . . . . . . . . . . . 179client.DisableHotWord configuration option . 180client.HotKeyBasePath configuration

option . . . . . . . . . . . . . 161, 181client.HotKeyLocalPath configuration

option . . . . . . . . . . . . . 161, 181client.ping.frequency configuration option . . 177client.ping.timeout configuration option . . . 177client.resource.address configuration

option . . . . . . . . . . . . . . . . 160client.resource.name configuration option . . 161client.resource.port configuration

option . . . . . . . . . . . . . 161, 179client.resource.uri configuration option 160, 180client.timeout configuration option . . . 120, 177cluster

High Availability . . . . . . . . . . . . . . 134Cluster Manager . . . . . . . . . . . . . . . 135

role in GVP . . . . . . . . . . . . . . . . . 37cluster mode, Resource Manager . . . . . . 136cluster_ip configuration option . . . . . 125, 136cmserviceagent configuration section . . . . 125codec configuration option. . . . . . . . 44, 165codecpref configuration option. . . . . . . . 166codecs

amr . . . . . . . . . . . . . . . . . . . . . 44g726. . . . . . . . . . . . . . . . . . . . . 44g729. . . . . . . . . . . . . . . . . . . . . 44gsm . . . . . . . . . . . . . . . . . . . . . 44h263. . . . . . . . . . . . . . . . . . . . . 44h263-1998 . . . . . . . . . . . . . . . . . 44media negotiation . . . . . . . . . . . . . . 43multiple . . . . . . . . . . . . . . . . . . . 44pcma . . . . . . . . . . . . . . . . . . . . 44pcmu . . . . . . . . . . . . . . . . . . . . 44supported . . . . . . . . . . . . . . . . . . 44telephone-event. . . . . . . . . . . . . . . 44tfci . . . . . . . . . . . . . . . . . . . . . . 44

com.genesyslab.var (metrics prefix) . . . . . .65commenting on this document . . . . . . . . .17Component ID . . . . . . . . . . . . . . . . .72components

Call Control Platform . . . . . . . . . . 22, 50configuring in Genesys Administrator . . . . 78

Index


Fetching Module. . . . . . . . . . . . . . . 22GVP . . . . . . . . . . . . . . . . . . . . . 21High Availability . . . . . . . . . . . . . . . 37identifiers . . . . . . . . . . . . . . . . . . 73Media Control Platform . . . . . . . . . 22, 38Reporting Server . . . . . . . . . . . . . . 23Resource Manager . . . . . . . . . . . . . 21Third-party Squid . . . . . . . . . . . . . . 22

conferenceallowed . . . . . . . . . . . . . . . . . . 143bridge . . . . . . . . . . . . . . . . . . . . 50configuring. . . . . . . . . . . . . . . . . 102events . . . . . . . . . . . . . . . . . . . 267ID . . . . . . . . . . . . . . . . . 34, 50, 151implicit . . . . . . . . . . . . . . . . . . . . 52input and output control . . . . . . . . . . . 43IVR Profile . . . . . . . . . . . . . . .103, 143limit . . . . . . . . . . . . . . . . . . . . 165NETANN requirements . . . . . . . . . . . 50selecting a resource . . . . . . . . . . . . . 34service selected . . . . . . . . . . . . . . . 31service, configuring . . . . . . . . . . . . 133services . . . . . . . . . . . . . . . . . . . 52services, configuring. . . . . . . . . . . . 102size . . . . . . . . . . . . . 35, 133, 142, 165

conference configuration section. . . . 103, 164conference-allowed configuration option 103, 143conference-capability-requirements

configuration option . . . . . . . . . 103conference-forbidden-respcode configuration

option . . . . . . . . . . . . . . . . 116conference-forbidden-set-alarm

configuration option . . . . . . . . . 116conference-id configuration option . . . 103, 151conference-sip-error-respcode

configuration option . . . . . . 103, 116conference-usage-limit configuration option . 103conference-usage-limit-exceeded-respcode

configuration option . . . . . . . . . 116conference-usage-limit-exceeded-set-alarm

configuration option . . . . . . . . . 116conference-usage-limit-per-session

configuration option . . . . . . . . . 103configuration file, device profile . . . . . . . 54configuration file, Squid . . . . . . . . . . . 59Configuration Layer . . . . . . . . . . . . . 77configuration options

<protocol>_proxy . . . . . . . . . . . . . 199<service>.<param- name> . . . . . . . . 150<service>-capability-requirement . . . . . 145<service>-forbidden-respcode. . . . . . . 146<service>-forbidden-set-alarm . . . . . . 146<service>-usage-limit . . . . . . . . . . . 146<service>-usage-limit-exceeded-

respcode . . . . . . . . . . . . . . . . 146

<service>-usage-limit-exceeded-set-alarm . . . . . . . . . . . . . . . . . . 147

<service>-usage-limit-per-session . . . . 147acceptcalltimeout . . . . . . . . . . . . . 120all . . . . . . . . . . . . . . . . . . . . . 112alternatevoicexml . . . . . . . . . . . . . 151aor . . . . . . . . . . . . . . . . . . . . 129appendrejcodec . . . . . . . . . . . . . . 165application-confmaxsize . . . . . . . 103, 142audio_format . . . . . . . . . . . . . . . 164bridge_server . . . . . . . . . . . . . . . 187bridge_server.profile . . . . . . . . . . . 187call-timeout . . . . . . . . . . . . . . . . 208capability . . . . . . . . . . . . . . . . . 131ccpccxml.default_uri . . . . . . . . . . . . 51ccxml-usage-limit-exceeded-respcode . . 116ccxml-usage-limit-exceeded-set-alarm . . 116client.ConnectPerSetup. . . . . . . . . . 179client.DisableHotWord . . . . . . . . . . 180client.HotKeyBasePath . . . . . . . . 161, 181client.HotKeyLocalPath . . . . . . . . 161, 181client.ping.frequency . . . . . . . . . . . 177client.ping.timeout. . . . . . . . . . . . . 177client.resource.address . . . . . . . . . . 160client.resource.name . . . . . . . . . . . 161client.resource.port . . . . . . . . . . 161, 179client.resource.uri . . . . . . . . . . . 160, 180client.timeout . . . . . . . . . . . . . 120, 177cluster_ip . . . . . . . . . . . . . . . 125, 136codec . . . . . . . . . . . . . . . . . 44, 165codecpref . . . . . . . . . . . . . . . . . 166conference-allowed . . . . . . . . . . 103, 143conference-capability-requirements. . . . 103conference-forbidden-respcode. . . . . . 116conference-forbidden-set-alarm . . . . . 116conference-id . . . . . . . . . . . . . 103, 151conference-sip-error-respcode . . . . 103, 116conference-usage-limit . . . . . . . . . . 103conference-usage-limit-exceeded-

respcode . . . . . . . . . . . . . . . . 116conference-usage-limit-exceeded-set-

alarm . . . . . . . . . . . . . . . . . . 116conference-usage-limit-per-session . . . 103confserver. . . . . . . . . . . . . . . 103, 167connect_timeout . . . . . . . . . . . . . . 95connection.timeout . . . . . . . . . . . . 120copy_unknown_headers . . . . . . . . . 117copyunknownheaders. . . . . . . . . . . 117dc.default.logfilter . . . . . . . . . . . . . 104dc.default.metricsfilter. . . . . . . . . . . 105debug . . . . . . . . . . . . . . . . . . . 112debug.enabled . . . . . . . . . . . . . . 178default.connecttimeout . . . . . . . . . . 120default_audio_format . . . . . . . . . . . 166default_uri . . . . . . . . . . . . . . . . . 186default-application . . . . . . . . . . 141, 154

Index

User’s Guide 325

defaultblindxfer . . . . . . . . . . . . 47, 167defaultbridgexfer. . . . . . . . . . . . 47, 167defaultconsultxfer . . . . . . . . . . . 47, 168defaultgw . . . . . . . . . . . . . . . . . 168defaulthost. . . . . . . . . . . . . . . . . 168default-properties-page . . . . . . . . . . 151defaultrejectcode . . . . . . . . . . . . . 117deferoutalerting . . . . . . . . . . . . . . 169dialing-rule-forbidden-respcode . . . .116, 143dialing-rule-forbidden-set-alarm . . . .116, 144dnis_correlationid_length . . . . . . . . . 169dnis_correlationid_offset . . . . . . . . . 169dsthours . . . . . . . . . . . . . . . . . . 215EMS Logging . . . . . . . . . . . . . . . 111ems.dc.default.metricsfilter . . . . . . . . . 65ems.ors.reportinginterval . . . . . . . . . . 68ems.rc.cdr.batch_size . . . . . . . . . . . . 68enablehttps . . . . . . . . . . . . . . 95, 215enablesendrecvevents . . . . . . . . . . 169expire . . . . . . . . . . . . . . . . . . . 113external-sip-allowed . . . . . . . . . . . . 144external-sip-forbidden-respcode. . . . . . 117external-sip-forbidden-set-alarm . . . . . 117external-sip-usage-limit-exceeded-set-

alarm . . . . . . . . . . . . . . . . . . 117fetch.timeout. . . . . . . . . . . . . . . . 121for conference . . . . . . . . . . . . . . . 103for logging . . . . . . . . . . . . . . . . . 112for session timers . . . . . . . . . . . . . 118full_<media type>_codec . . . . . . . . . 188hfdisctimer . . . . . . . . . . . . . . .120, 170hfprefix. . . . . . . . . . . . . . . . . . . 170hfstopdial . . . . . . . . . . . . . . . . . 170hftype . . . . . . . . . . . . . . . . . . . 170https_proxy . . . . . . . . . . . . . . . . 100httptimeout. . . . . . . . . . . . . . . . . 215in.<SIP request>.headers . . . . . . . . . 171in.invite.params . . . . . . . . . . . . . . 171inbound_allowed_media. . . . . . . . . . 188inbound-usage-limit . . . . . . . . . . . . 144inbound-usage-limit-exceeded-set-alarm . 117info.contenttype . . . . . . . . . . . . . . 171initial_request_fetchtimeout . . . . . . . . 120initial_request_method . . . . . . . . . . 178initial-page-url . . . . . . . . . . . . . . . 151interaction . . . . . . . . . . . . . . . . . 112IVR Profile, for conference . . . . . . . . 103limit . . . . . . . . . . . . . . . . . . . . 165load_once_per_call . . . . . . . . . .144, 164load-balance-scheme . . . . . . . . . . . 131localrtpaddr . . . . . . . . . . . . . . . . 172localtimeformat . . . . . . . . . . . . . . 215log_sinks . . . . . . . . . . . . . . . . . 105logconfig.<Sink Name> . . . . . . . . . . 106max_script_time . . . . . . . . . . . . . . 121maxincalltime . . . . . . . . . . . . . . . 120

max-page-count. . . . . . . . . . . . . . 208max-page-size . . . . . . . . . . . . . . 209mcp-asr-usage-mode . . . . . . . . . . . 144mcp-max-log-level . . . . . . . . . . . . 144mcp-sendrecv-enabled . . . . . . . . . . 145message_format . . . . . . . . . . . . . 113metadata . . . . . . . . . . . . . . . . . . 81metricsconfig.<Sink Name>. . . . . . . . 107metricsfilter . . . . . . . . . . . . . . . . 143min_se . . . . . . . . . . . . . . . . . . 119monitor-method . . . . . . . . . . . . . . 132no_cache_url_substr . . . . . . . . . . . 199noresource-response-code . 116, 130, 132, 287OPTIONS.header.Accept . . . . . . . . . 117OPTIONS.header.Accept-Encoding . . . 117OPTIONS.header.Accept-Language . . . 117OPTIONS.header.Allow. . . . . . . . . . 117options_response_contenttype . . . . . . 116options_response_msg_body. . . . . . . 116ors.reportinginterval . . . . . . . . . . . . 107out.<SIP request>.headers . . . . . . . . 172out.<SIP request>.params . . . . . . . . 172outbound-call-allowed. . . . . . . . . . . 145outbound-call-forbidden-respcode . . . . 117outbound-call-forbidden-set-alarm . . . . 117outbound-usage-limit . . . . . . . . . . . 145outbound-usage-limit-exceeded-set-

alarm . . . . . . . . . . . . . . . . . . 117outcalluseoriggw . . . . . . . . . . . . . 173platform.save_<file type>_files . . . . . . 187port-capacity . . . . . . . . . . . . . . . 130port-usage-type . . . . . . . . . . . . 34, 133quartz.rs.calltimeout . . . . . . . . . . . 210quartz.rs.dbMaintenancePeriod. . . . . . 210rc.batch_size . . . . . . . . . . . . . . . 107rc.cdr.batch_size . . . . . . . . . . . . . 108rc.cdr.local_queue_path. . . . . . . . . . 108rc.cdr.msg_broker_uri . . . . . . . . . . . 108rc.local_queue_path . . . . . . . . . . . 109rc.msg_broker_uri. . . . . . . . . . . . . 109rc.ors.msg_broker_uri. . . . . . . . . . . 109rc.ors_local_queue_path . . . . . . . . . 109referxferhold . . . . . . . . . . . . . . . 173registration . . . . . . . . . . . . . . . . 174reporting . . . . . . . . . . . . . . . . . 104resource-confmaxcount . . . . . . . . . . 134resource-confmaxsize . . . . . . . . . . 133resource-no-match-respcode . . . . . . . 116resource-unavailable-respcode . . . . . . 116route.default.<protocol>. . . . . . . . . . . 88route.dest.<n> . . . . . . . . . . . . . 87, 89routeset . . . . . . . . . . . . . . . . . 87, 90rs.db.retention.<type of data>.default . . . 209rs.httpport . . . . . . . . . . . . . . . . . 215rs.password . . . . . . . . . . . . . . . . 215rs.query.limit.<granularity> . . . . . . . . 205

Index


rs.query.limit.<time period> . . . . . . . . 210rs.username . . . . . . . . . . . . . . . . 216rtp.timeout . . . . . . . . . . . . . . . . . 120rule-<n> . . . . . . . . . . . . . . . . . . 149securerouteset. . . . . . . . . . . . . . 87, 91segment . . . . . . . . . . . . . . . . . . 114sendalert. . . . . . . . . . . . . . . .117, 175service-type . . . . . . . . . . . . . .103, 142service-types . . . . . . . . . . . . . . . 133sessionexpires. . . . . . . . . . . . . . . 119sessionparams . . . . . . . . . . . . . . . 95sessionparamsoffer . . . . . . . . . . . . . 95setting dynamically . . . . . . . . . . . . 295sip.allowedunknownheaders . . . . . . . 186sip.min_se . . . . . . . . . . . . . . . . . 119sip.proxy.optionsinterval . . . . . . . . . . 126sip.proxy.unavailoptionsinterval . . . . . . 126sip.registrar.maxexpirytime . . . . . . . . 120sip.registrar.minexpirytime. . . . . . . . . 120sip.route.default.<protocol> . . . . . . . . . 88sip.route.dest.<n> . . . . . . . . . . . . . . 89sip.route.dests . . . . . . . . . . . . . . . . 90sip.send_progressing . . . . . . . . . . . 117sip.sessionexpires . . . . . . . . . . . . . 119sip.sessiontimer . . . . . . . . . 118, 142, 155sip.timer.ci_proceeding . . . . . . . . . . 120sip.transport.<x>. . . . . . . . . . 86, 92, 136sip-header-for-dnis . . . . . . . . . . 30, 126sipinfoallowedcontenttypes . . . . . . . . 175sipinfodtmf . . . . . . . . . . . . . . . . . . 41sipproxy . . . . . . . . . . . . . . . . . . 188srtp.cryptomethods . . . . . . . . . . . . . 95srtp.mode . . . . . . . . . . . . . . . 95, 166ssl_* . . . . . . . . . . . . . . . . . . . . 199ssl_cipher_list . . . . . . . . . . . . . . . 101ssl_key . . . . . . . . . . . . . . . . . . 100ssl_key_passwd . . . . . . . . . . . . . . 100standard . . . . . . . . . . . . . . . . . . 112suspend-mode-respcode . . . . . . . . . 116time-format . . . . . . . . . . . . . . . . 114trace . . . . . . . . . . . . . . . . . . . . 112trace_flag . . . . . . . . . . . . . . . . . 110transfer.allowed . . . . . . . . . . . . . . 178transfer-allowed . . . . . . . . . . . . . . 147transfer-forbidden-respcode . . . . . . . . 117transfer-forbidden-set-alarm. . . . . . . . 117transfermethods . . . . . . . . . . . . . . 175transport.<x> . . . . . . . . . . . . . . . . 92tzoffset. . . . . . . . . . . . . . . . . . . 216usage-limit-exceeded-respcode . . . .117, 147usage-limit-exceeded-set-alarm. . . .117, 147usage-limits . . . . . . . . . . . . . .147, 155userdata.prefix. . . . . . . . . . . . . . . 178use-same-gateway . . . . . . . . . . . . 148verbose . . . . . . . . . . . . . . . . . . 115voicexml-dialog-allowed . . . . . . . . . . 148

voicexml-dialog-forbidden-respcode . . . 117voicexml-dialog-forbidden-set-alarm . . . 117voicexml-usage-limit-exceeded-

respcode . . . . . . . . . . . . . . . . 117voicexml-usage-limit-exceeded-set-

alarm . . . . . . . . . . . . . . . . . . 117vrm.client.TlsCertificateKey . . . . . . . . . 95vrm.client.TlsPassword . . . . . . . . . . . 95vrm.client.TlsPrivateKey . . . . . . . . . . 95vxmli.initial_request_maxage . . . . . . . . 58vxmli.initial_request_maxstale . . . . . . . 58vxmlinvite . . . . . . . . . . . . . . . . . 176warningheaders . . . . . . . . . . . . . . 176xfer.copyheaders . . . . . . . . . . . . . 176

configuration sectionsasr. . . . . . . . . . . . . . . . . . . . . 164calllog . . . . . . . . . . . . . . . . . . . 162ccpccxml . . . . . . . . . . . . . . . 185, 186ccxmli . . . . . . . . . . . . . . . . . 185, 187cdr. . . . . . . . . . . . . . . . . . . . . 208cmserviceagent . . . . . . . . . . . . . . 125conference . . . . . . . . . . . . . . 103, 164dbmp . . . . . . . . . . . . . . . . . 140, 209e-mail . . . . . . . . . . . . . . . . . . . 162ems . . . . . . . . . . . . . . . . . . . . 104gvp.dnis-range . . . . . . . . . . . . . . 154gvp.general . . . . . . . . . . . . . . 141, 154gvp.log . . . . . . . . . . . . . . . . . . 142gvp.policy . . . . . . . . . . . . . . . 143, 155gvp.policy.dialing-rules . . . . . . . . . . 148gvp.service-parameters . . . . . . . . 149, 155gvp.service-prerequisite . . . . . . . . . 150iproxy . . . . . . . . . . . . . . . . . . . 199log. . . . . . . . . . . . . . . . . . . . . 112mediacontroller . . . . . . . . . 103, 186, 187mediactrller . . . . . . . . . . . . . . 103, 186messaging . . . . . . . . . . . . . . . . 207monitor . . . . . . . . . . . . . . 92, 125, 126mpc . . . . . . . . . . . . . . . . . . . . 165mtinternal . . . . . . . . . . . . . . . . . 162persistence . . . . . . . . . . . . . . . . 207proxy . . . . . . . . . . . . . . . . . 92, 125registrar . . . . . . . . . . . . . . . . 92, 125reporting . . . . . . . . . . . . . . . 207, 210rm . . . . . . . . . . . . . . . . . . . . . 125schedule . . . . . . . . . . . . . . . 207, 210session . . . . . . . . . . . . . . . . . . 186sessmgr . . . . . . . . . . . . . . . . 95, 162sip . . . . . . . . . . . . 92, 95, 163, 167, 186stack . . . . . . . . . . . . . . . . . . . 163transaction . . . . . . . . . . . . . . . . 207vrm . . . . . . . . . . . . . . . . . . 163, 179vxmli . . . . . . . . . . . . . . . . . 163, 177

Configuration Server data . . . . . . . . . . .73configuring

access control for reporting . . . . . . . . 211

Index

User’s Guide 327

ASR and TTS . . . . . . . . . . . . . . . 159Call Control Platform . . . . . . . . . . . 184CCXML service . . . . . . . . . . . . . . 133conference service . . . . . . . . . .102, 133database retention policies . . . . . .140, 206default IVR Profile . . . . . . . . . . . . . 140device profiles . . . . . . . . . . . . . . . 188DNIS mapping . . . . . . . . . . . . . . . 152external SIP service . . . . . . . . . . . . 133Fetching Module. . . . . . . . . . . . . . 198Fetching Module for HTTPS. . . . . . . . . 99gateway service . . . . . . . . . . . . . . 133GVP . . . . . . . . . . . . . . . . . . . . . 82GVP for SIP Server integration . . . . . . 321High Availability . . . . . . . . . . . . . . 134in Genesys Administrator . . . . . . . . . . 78IVR Profile policies . . . . . . . . . . . . 143load-balancing . . . . . . . . . . . . . . . 131Media Control Platform . . . . . . . . . . 158options in the Genesys Administrator . . . . 81Reporting Server . . . . . . . . . . . . . 203Reporting Server connection to Genesys

Administrator . . . . . . . . . . . . . . 213resource groups . . . . . . . . . . . .126, 127Resource Manager . . . . . . . . . . . . 124Resource Manager in cluster mode . . . . 136route set . . . . . . . . . . . . . . . . . . . 87safe ports, Squid . . . . . . . . . . . . . 202service types . . . . . . . . . . . . .133, 142SIP communications and routing . . . . . . 86Squid . . . . . . . . . . . . . . . . . . . 200SSL ports, Squid . . . . . . . . . . . . . 202Tomcat for SSL . . . . . . . . . . . . . . 101VoiceXML service . . . . . . . . . . . . . 133

confmaxsize parameter . . . . . . . . . .34, 35confreserve parameter . . . . . . . . . . .34, 35confserver configuration option . . . . 103, 167connect_timeout configuration option . . . . 95connection

events . . . . . . . . . . . . . . . . . . . 267outgoing . . . . . . . . . . . . . . . . . . . 53

connection.disconnect.hangup event . . . . 42connection.timeout configuration option . . . 120consultation transfer type . . . . . . . . .45, 47control sessions

MRCPv1. . . . . . . . . . . . . . . . . . . 44MRCPv2. . . . . . . . . . . . . . . . . . . 44

controllingaccess for reporting . . . . . . . . . . . . . 69

Convedia Media Server (device profile) . 54, 305copy_unknown_headers configuration

option . . . . . . . . . . . . . . . . 117copyunknownheaders configuration option . 117CPA support . . . . . . . . . . . . . . . . . 43createcall . . . . . . . . . . . . . . . . . . . 53create-CCXML (CDR call type) . . . . . . . 67

creatingSSL private key and certificate . . . . . . . 96SSL private key and self-signed certificate . 97

credentialsfor reporting . . . . . . . . . . . . . . . . . 69

CTI userdata . . . . . . . . . . . . . . . . . .42custom

device profiles. . . . . . . . . . . . . . . 193SIP headers. . . . . . . . . . . . . . . 40, 42SIP responses . . . . . . . . . 116, 117, 130

DData Collection Sink

See DATACdatabase, reporting

default retention periods . . . . . . . . . 206maintenance process . . . . . . . . . . . . 69purging . . . . . . . . . . . . . . . . . . . 69retention policies . . . . . . . . . . . 140, 206

DATACdescribed . . . . . . . . . . . . . . . . . . 65metrics filter . . . . . . . . . . . . . . . . . 65role and functioning . . . . . . . . . . . . . 65statistics . . . . . . . . . . . . . . . . . . . 66

dbmp configuration section . . . . . . 140, 209dc.default.logfilter configuration option . . . 104dc.default.metricsfilter configuration option . 105debug configuration option . . . . . . . . . 112debug.enabled configuration option . . . . . 178debugging

CCXML applications . . . . . . . . . . . . 53Media Control Platform operations . . . . 178platform operations . . . . . . . . . . . . . 50VoiceXML applications . . . . . . . . . . . 50

defaultCCXML application URI. . . . . . . . . . 186database retention periods . . . . . . 206, 209device profiles. . . . . . . . . . . . . 54, 304gateway . . . . . . . . . . . . . . . . . . 168IVR Profile . . . . . . . . . . . . . . 140, 154log filters . . . . . . . . . . . . . . . . . 110log option values . . . . . . . . . . . . . 115metrics filters . . . . . . . . . . . . . . . 110SIP transports . . . . . . . . . . . . . . . . 93SSL private key and certificate paths . . . . 97transfer methods . . . . . . . . . . . . . 167

default application (Configuration Manager) . 214Default Conference (device profile) . . . 54, 305Default Dialog (device profile) . . . . . . 54, 305Default Inbound (device profile) . . . . . 54, 305Default Outbound (device profile) . . . . 54, 305default.connecttimeout configuration

option . . . . . . . . . . . . . . . . 120default_audio_format configuration option. . 166default_uri configuration option . . . . . . . 186

Index


default-application configuration option . . . . . . . . . . . . . 141, 154

defaultblindxfer configuration option . . . 47, 167defaultbridgexfer configuration option . . 47, 167defaultconsultxfer configuration option . . 47, 168defaultgw configuration option . . . . . . . . 168defaulthost configuration option . . . . . . . 168default-properties-page configuration option . 151defaultrejectcode configuration option . . . . 117deferoutalerting configuration option . . . . . 169deleting resource groups . . . . . . . . . . . 129delivery

batch, of CDRs and metrics . . . . . . . . . 68of CDRs . . . . . . . . . . . . . . . . . . . 66of logs . . . . . . . . . . . . . . . . . . . . 64of metrics . . . . . . . . . . . . . . . . . . 65OR statistics . . . . . . . . . . . . . . . . . 68Reporting Client . . . . . . . . . . . . . . . 68

dest attribute . . . . . . . . . . . . . . . . . 53device profiles

assigning . . . . . . . . . . . . . . . . . . 54configuration file . . . . . . . . . . . . 54, 189configuring. . . . . . . . . . . . . . . . . 188customizing . . . . . . . . . . . . . . . . 193default . . . . . . . . . . . . . . . . . 54, 304defined. . . . . . . . . . . . . . . . . . . . 53properties . . . . . . . . . . . . . . . . . 189provisioning . . . . . . . . . . . . . .188, 194

dialed number mapping . . . . . . . . 151, 154dialing rules . . . . . . . . . . . . . . . . . 148dialing-rule-forbidden-respcode

configuration option . . . . . . 116, 143dialing-rule-forbidden-set-alarm

configuration option . . . . . . 116, 144dialog

events . . . . . . . . . . . . . . . . . . . 267media-less . . . . . . . . . . . . . . . . . . 41MRCP client/server . . . . . . . . . . . . . 44

dialogsinitiating transfers . . . . . . . . . . . 52, 178initiating transfers and conferences . . . . . 42VoiceXML . . . . . . . . . . . . . . . . . . 42

Display Filter, on Settings tab . . . . . . . . 80DNIS identification . . . . . . . . . . . . . . 30DNIS, mapping IVR Profiles . . . . . . . . . 151dnis_correlationid_length configuration

option . . . . . . . . . . . . . . . . 169dnis_correlationid_offset configuration

option . . . . . . . . . . . . . . . . 169document

commenting on . . . . . . . . . . . . . . . 17conventions . . . . . . . . . . . . . . . . . 13errors, commenting on . . . . . . . . . . . 15version number . . . . . . . . . . . . . . . 13

dsthours configuration option . . . . . . . . 215

DTMFaccept digits. . . . . . . . . . . . . . . . . 41audio files . . . . . . . . . . . . . . . . . . 41barge-in . . . . . . . . . . . . . . . . . . . 43event . . . . . . . . . . . . . . . . . . 41, 52grammars . . . . . . . . . . . . . . . . . . 45hookflash transfer . . . . . . . . . . . . 46, 47navigation . . . . . . . . . . . . . . . . . . 43recording input . . . . . . . . . . . . . . . 43

dynamic configuration options . . . . . . . . 295

Eemail configuration section . . . . . . . . . 162ems configuration section . . . . . . . . . . 104EMS Logging

configuration options . . . . . . . . . . . 111logs . . . . . . . . . . . . . . . . . . . . . 63metrics . . . . . . . . . . . . . . . . . . . 63role and functioning . . . . . . . . . . . 62, 63

EMS Logging interface. . . . . . . . . . . . .62EMS Reporting

configuration options . . . . . . . . . . . 104See also CDRs, logs, metrics, OR reporting,

reportingEMS Reporting, defined . . . . . . . . . . . .60ems.dc.default.metricsfilter configuration

option . . . . . . . . . . . . . . . . .65ems.ors.reportinginterval configuration option.68ems.rc.cdr.batch_size configuration option . .68enablehttps configuration option . . . . . 95, 215enablesendrecvevents configuration option . 169enabling

ASR . . . . . . . . . . . . . . . . . . . . 159debugging. . . . . . . . . . . . . . . . . 178High Availability . . . . . . . . . . . . . . 134HTTP Basic Authorization for reporting . . 211HTTPS for reporting . . . . . . . . . . . 215reporting . . . . . . . . . . . . . . . . . . 82SIPS, HTTPS, SRTP . . . . . . . . . . . . 94SRTP . . . . . . . . . . . . . . . . . . . . 95TTS . . . . . . . . . . . . . . . . . . . . 159

encoding . . . . . . . . . . . . . . . . . . 39, 40encoding attribute . . . . . . . . . . . . . . .41encryption

SRTP keys and options . . . . . . . . . . . 71enctype parameter . . . . . . . . . . . . . . .52enforcing policies . . . . . . . . . . . . . . .32Environment tenant

IVR Profile settings . . . . . . . . . . . . 153session timers. . . . . . . . . . . . . . . 118

errors . . . . . . . . . . . . . . . . . . . . .17events

conference . . . . . . . . . . . . . . . . 267connection . . . . . . . . . . . . . . . . 267connection.disconnect.hangup . . . . . . . 42

Index

User’s Guide 329

dialog . . . . . . . . . . . . . . . . . . . 267DTMF . . . . . . . . . . . . . . . . . . 41, 52logs . . . . . . . . . . . . . . . . . . . . . 64media controller . . . . . . . . . . . . . . 269

expire configuration option . . . . . . . . . . 113Expires header . . . . . . . . . . . . . . . . 56Expires headers . . . . . . . . . . . . . .57, 59expiry timers . . . . . . . . . . . . . . . . . 118exponential back-off . . . . . . . . . . . . . 68exporting Configuration Server data . . . . . 73external (CDR call type) . . . . . . . . . . . 67external SIP service . . . . . . . . . . . . . 32external-sip

service, configuring . . . . . . . . . . . . 133external-sip-allowed configuration option . . 144external-sip-forbidden-respcode

configuration option . . . . . . . . . 117external-sip-forbidden-set-alarm

configuration option . . . . . . . . . 117external-sip-usage-limit-exceeded-set-

alarm configuration option . . . . . . 117

Ffailures

gateway . . . . . . . . . . . . . . . .130, 132Resource Manager handling . . . . . 36, 130

Far-East encoding . . . . . . . . . . . . . . 41fetch

data . . . . . . . . . . . . . . . . . . . . . 53parameters . . . . . . . . . . . . . . . . . 52

fetch methods . . . . . . . . . . . . . . . . 52fetch.timeout configuration option . . . . . . 121Fetching Module

and upstream metrics . . . . . . . . . . . 104caching . . . . . . . . . . . . . . . . . 55, 56caching algorithm . . . . . . . . . . . . . 315configuring. . . . . . . . . . . . . . . . . 198configuring for HTTPS. . . . . . . . . . . . 99described . . . . . . . . . . . . . . . . . . 55enabling secure communications . . . . . . 94functions . . . . . . . . . . . . . . . . . . . 22HTTPS. . . . . . . . . . . . . . . . . . . . 94module IDs . . . . . . . . . . . . . . . . 276role in call flow. . . . . . . . . . . . . . . . 26shared memory . . . . . . . . . . . 38, 51, 55SNMP . . . . . . . . . . . . . . . . . . . . 70specifier IDs . . . . . . . . . . . . . . . . 276SSL configuration . . . . . . . . . . . . . 199start order . . . . . . . . . . . . . . . . . . 55

file formatsaudio, play . . . . . . . . . . . . . . . . . 289audio, record . . . . . . . . . . . . . . . 292audio/video, play . . . . . . . . . . . . . 291audio/video, record . . . . . . . . . . . . 294video, play . . . . . . . . . . . . . . . . . 291

video, record . . . . . . . . . . . . . . . 293files

device profile configuration . . . . . . . . 189hotkey grammar. . . . . . . . . . . . . . 161log. . . . . . . . . . . . . . . . . . . . . . 64

filtersdefault for logs and metrics . . . . . . . . 110for metrics delivery . . . . . . . . . . . . . 65for Settings tab display . . . . . . . . . . . 80logs . . . . . . . . . . . . . . . . . . 104, 106metrics . . . . . . . . . . . . . . . . 105, 107

full_<media type>_codec configuration option . . . . . . . . . . . . . . . . 188

Gg726 . . . . . . . . . . . . . . . . . . . . . .44g729 . . . . . . . . . . . . . . . . . . . . . .44gateway

default . . . . . . . . . . . . . . . . . . . 168failure responses . . . . . . . . . . . . . 132load-balancing . . . . . . . . . . . . . . . 34resources . . . . . . . . . . . . . . . . . 129response to failures . . . . . . . . . . . . 130service selected. . . . . . . . . . . . . . . 31service, configuring . . . . . . . . . . . . 133

Genesys Administratoraccessing . . . . . . . . . . . . . . . . . . 24configuration options metadata . . . . . . . 81configuring HTTPS . . . . . . . . . . . . . 95configuring objects . . . . . . . . . . . . . 78configuring options . . . . . . . . . . . . . 81connecting to Reporting Server . . . . . . 213described . . . . . . . . . . . . . . . . . . 23monitoring GVP . . . . . . . . . . . . . . 219more information . . . . . . . . . . . . . . 24Settings tab . . . . . . . . . . . . . . . 78, 79Settings tab Display Filter . . . . . . . . . . 80using . . . . . . . . . . . . . . . . . . . . 78

Genesys CallUUID. . . . . . . . . . . . . . .72Genesys Configuration Layer . . . . . . . . .77get if modified . . . . . . . . . . . . . . . . .57get method. . . . . . . . . . . . . . . . . . .52grammars

and HTTPS . . . . . . . . . . . . . . . . . 71builtin . . . . . . . . . . . . . . . . . . . . 45DTMF . . . . . . . . . . . . . . . . . . . . 45hotkey . . . . . . . . . . . . . . . . . 44, 161implied . . . . . . . . . . . . . . . . . . . 45inline . . . . . . . . . . . . . . . . . . . . 45

granularity . . . . . . . . . . . . . . . . . . 204groups, resource . . . . . . . . . . . . 126, 127gsm . . . . . . . . . . . . . . . . . . . . . .44GVP

component identifiers . . . . . . . . . . . . 73Component ID . . . . . . . . . . . . . . . 72

Index


components . . . . . . . . . . . . . . . . . 21configuring. . . . . . . . . . . . . . . . . . 82enabling ASR and TTS . . . . . . . . . . 159enabling reporting . . . . . . . . . . . . . . 82High Availability . . . . . . . . . . . . . . . 37HTTPS support . . . . . . . . . . . . . . . 70log levels . . . . . . . . . . . . . . . . . 104Manage Resources wizard . . . . . . . . 128MIBs . . . . . . . . . . . . . . . . . . . . . 23MRCP speech servers . . . . . . . . . . 160provisioning . . . . . . . . . . . . . . . . . 82sample call flow . . . . . . . . . . . . . . . 25secure communications . . . . . . . . . . . 70Session ID . . . . . . . . . . . . . . . . 28, 72SIP response codes . . . . . . . . . . . . 279SIPS support . . . . . . . . . . . . . . . . 70SNMP support . . . . . . . . . . . . . . . . 70SRTP support . . . . . . . . . . . . . . . . 71SSL support . . . . . . . . . . . . . . . . . 70TLS support . . . . . . . . . . . . . . . . . 70traps . . . . . . . . . . . . . . . . . . . . . 70

GVP MCP (device profile) . . . . . . . . 54, 305gvp.alternatevoicexml parameter . . . . . . 39gvp.config parameter. . . . . . . . . . . 39, 294gvp.dnis-range configuration section. . . . . 154gvp.general configuration section . . . 141, 154gvp.log configuration section . . . . . . . . . 142gvp.policy configuration section . . . . 143, 155gvp.policy parameters . . . . . . . . . . . . 40gvp.policy.dialing-rules configuration

section . . . . . . . . . . . . . . . . 148gvp.rm.datanode parameter . . . . . . . . . 28gvp.rm.tenant-id parameter . . . . . . . .28, 73gvp.rm-resource-req parameters. . . . . . . 34gvp.service-parameters configuration

section . . . . . . . . . . . . . 149, 155gvp.service-prerequisite configuration

section . . . . . . . . . . . . . . . . 150gvp-tenant-id parameter . . . . . . . . . . . 73

Hh263 . . . . . . . . . . . . . . . . . . . . . 44h263-1998 . . . . . . . . . . . . . . . . . . 44headers

Cache-Control . . . . . . . . . . . . 56, 57, 59Expires . . . . . . . . . . . . . . . 56, 57, 59If-Modified-Since (IMS) . . . . . . . . . . . 57Session-Expires . . . . . . . . . . . . 118, 119

health, of resources . . . . . . . . . . . . . 33hfdisctimer configuration option . . . . 120, 170hfprefix configuration option . . . . . . . . . 170hfstopdial configuration option . . . . . . . . 170hftype configuration option . . . . . . . . . . 170High Availability

cluster . . . . . . . . . . . . . . . . . . . 134

cluster mode . . . . . . . . . . . . . . . 136cmserviceagent section . . . . . . . . . . 125configuring . . . . . . . . . . . . . . . . 134enabling . . . . . . . . . . . . . . . . . . 134support in GVP . . . . . . . . . . . . . . . 37

hintsattribute . . . . . . . . . . . . . . . . . . 117customizing SIP responses . . . . . . . . 117outboundproxy . . . . . . . . . . . . . . . 53

historicalcall browser report . . . . . . . . . . . . 238call summary report . . . . . . . . . . . . 235peaks report. . . . . . . . . . . . . . . . 236reports. . . . . . . . . . . . . . . . . . . 233

HKF transfer . . . . . . . . . . . . . . . . 46, 47hookflash transfer . . . . . . . . 46, 47, 120, 170hotkey grammars . . . . . . . . . . . . 44, 161hotwords . . . . . . . . . . . . . . . . . . . .44HTTP

Basic Authorization . . . . . . . . . . . . 211fetch methods . . . . . . . . . . . . . . . . 52

HTTP/1.1-compliant caching. . . . . . . . . .56HTTPS

and caching . . . . . . . . . . . . . . . . . 71and Genesys Administrator . . . . . . . . . 95and grammars . . . . . . . . . . . . . . . 71and Reporting Server web server. . . . . . 95and VoiceXML applications . . . . . . . . . 71configuring Fetching Module . . . . . . . . 99considerations . . . . . . . . . . . . . . . 71enabling . . . . . . . . . . . . . . . . . . . 94enabling for reporting . . . . . . . . . . . 215fetch methods . . . . . . . . . . . . . . . . 52setting up Fetching Module . . . . . . . . . 94support . . . . . . . . . . . . . . . . . 52, 70

https_proxy configuration option . . . . . . . 100httptimeout configuration option . . . . . . . 215

Iidentifiers

application DBID . . . . . . . . . . . . . . 73conference . . . . . . . . . . . . . . . . 151for GVP applications . . . . . . . . . . . . 73for GVP components . . . . . . . . . . . . 73for GVP sessions . . . . . . . . . . . . . . 72Genesys CallUUID . . . . . . . . . . . . . 72GVP Component ID. . . . . . . . . . . . . 72GVP Session ID. . . . . . . . . . . . . . . 72IVR Profile . . . . . . . . . . . . . . 73, 140IVR Profile DBID . . . . . . . . . . . . . . 73module IDs, listed . . . . . . . . . . . . . 255specifier IDs, listed . . . . . . . . . . . . 255

If-Modified-Since headers . . . . . . . . . . .57IIS

and SSL. . . . . . . . . . . . . . . . . . . 97

Index

User’s Guide 331

and Tomcat . . . . . . . . . . . . . . . . . 95IIS application server

and grammars . . . . . . . . . . . . . . . . 45implementing

secure communications . . . . . . . . . . . 71implicit

conferencing. . . . . . . . . . . . . . . . . 52transcoding . . . . . . . . . . . . . . . . . 52

implied grammars . . . . . . . . . . . . . . 45importing Configuration Server data . . . . . 73IMS headers . . . . . . . . . . . . . . . . . 57in.<SIP request>.headers configuration

option . . . . . . . . . . . . . . . . 171in.invite.params configuration option . . . . . 171inactivity timers . . . . . . . . . . . . . . . . 118inbound (CDR call type) . . . . . . . . . . . 66inbound_allowed_media configuration

option . . . . . . . . . . . . . . . . 188inbound-usage-limit configuration option. . . 144inbound-usage-limit-exceeded-set-alarm

configuration option . . . . . . . . . 117INFO messages . . . . . . . . . . . . . . . 41info.contenttype configuration option. . . . . 171init.bat file . . . . . . . . . . . . . . . . . . 136initial page, caching . . . . . . . . . . . .52, 58initial_request_fetchtimeout configuration

option . . . . . . . . . . . . . . . . 120initial_request_method configuration option . 178initial-page-url configuration option. . . . . . 151inline grammars . . . . . . . . . . . . . . . 45input control, conference . . . . . . . . . . . 43intended audience . . . . . . . . . . . . . . 12interaction configuration option. . . . . . . . 112interfaces

<log> tag . . . . . . . . . . . . . . . . . . 63CDR service . . . . . . . . . . . . . . . . . 62logging. . . . . . . . . . . . . . . . . . . . 62OR. . . . . . . . . . . . . . . . . . . . . . 68OR service. . . . . . . . . . . . . . . . . . 62VAR . . . . . . . . . . . . . . . . . . . . . 63

ioproc. . . . . . . . . . . . . . . . . . . . . 50iproxy configuration section . . . . . . . . . 199ISO-8859-x . . . . . . . . . . . . . . . . . . 41italics . . . . . . . . . . . . . . . . . . . . . 14ivr action usage report . . . . . . . . . . . . 248IVR Profile

assigning default to tenant . . . . . . . . 154configuring. . . . . . . . . . . . . . . . . . 78configuring DNIS mapping . . . . . . . . 152customizing SIP responses . . . . . . . . 116database retention policies . . . . . .140, 206DBID. . . . . . . . . . . . . . . . . . . 28, 73default . . . . . . . . . . . . . . . . .140, 154defined. . . . . . . . . . . . . . . . . 23, 139dialed number mapping . . . . . . . . . . 151for conference . . . . . . . . . . . . . . . 103

mapping calls to. . . . . . . . . . .30, 77, 151metrics filter . . . . . . . . . . . . . . . . 143name . . . . . . . . . . . . . . . . . 73, 140policies . . . . . . . . . . . . . . . . 28, 143provisioning . . . . . . . . . . . . . . . . 139selecting . . . . . . . . . . . . . . . . . . 30session timers. . . . . . . . . . . . . . . 118tenant settings . . . . . . . . . . . . . . 153

JJapanese, encoding for . . . . . . . . . . . .41JMS Message Broker . . . . . . . . . . . . 109join-style transfer. . . . . . . . . . . . . . . .46

KKorean, encoding for . . . . . . . . . . . . .41

Llast ivr action used report . . . . . . . . . . 249least used load balancing . . . . . . . . . . 131least used load-balancing . . . . . . . . . . .34limit configuration option . . . . . . . . . . . 165limits, conference size . . . . . . . . . . . . 165load_once_per_call configuration

option . . . . . . . . . . . . . 144, 164load-balance-scheme configuration option . 131load-balancing

configuring . . . . . . . . . . . . . . . . 131for gateway service . . . . . . . . . . . . . 34least used . . . . . . . . . . . . . . . 34, 131MRCP services . . . . . . . . . . . . . . . 34NLB clusters . . . . . . . . . . . . . . . . 37round robin . . . . . . . . . . . . . . 34, 131service requests. . . . . . . . . . . . . . . 34

localrtpaddr configuration option. . . . . . . 172localtimeformat configuration option . . . . . 215log configuration section . . . . . . . . . . . 112log sinks . . . . . . . . . . . . . . . . . . 64, 65

DATAC . . . . . . . . . . . . . . . . . . . 65default . . . . . . . . . . . . . . . . . . . 105described . . . . . . . . . . . . . . . . . . 65MFSINK . . . . . . . . . . . . . . . . . . . 65TRAPSINK . . . . . . . . . . . . . . . . . 66

log_sinks configuration option . . . . . . . . 105logconfig.<Sink Name> configuration

option . . . . . . . . . . . . . . . . 106logs

caching proxy . . . . . . . . . . . . . . . . 60configuration options . . . . . . . . . . . 112data . . . . . . . . . . . . . . . . . . . . . 64default configuration values. . . . . . . . 115

Index


default filters . . . . . . . . . . . . . . . . 110defined. . . . . . . . . . . . . . . . . . . . 63delivery . . . . . . . . . . . . . . . . . . . 64described . . . . . . . . . . . . . . . . . . 63file location . . . . . . . . . . . . . . . . . 64filter . . . . . . . . . . . . . . . . . .104, 106generated . . . . . . . . . . . . . . . . 42, 53levels . . . . . . . . . . . . . . . . . . . 104logging interface . . . . . . . . . . . . . . . 62module IDs . . . . . . . . . . . . . . . . . 64See also log sinksseverity . . . . . . . . . . . . . . . . . . . 63specifiers . . . . . . . . . . . . . . . . . . 64Squid . . . . . . . . . . . . . . . . . 60, 317

lr parameter . . . . . . . . . . . . . . . . . 32

MManagement Framework. . . . . . . . . . . 33Management Framework Adaptation Sink

See MFSINKmanaging

resources . . . . . . . . . . . . . . . . . . 32sessions . . . . . . . . . . . . . . . . . . 118Squid cache . . . . . . . . . . . . . . . . . 60

manual cache management . . . . . . . . . 60mapping

calls to IVR Profiles . . . . . . . . . . 30, 151configuring DNIS . . . . . . . . . . . . . 152dialed numbers . . . . . . . . . . . . . . 154

Master Agent (for SNMP) . . . . . . . . . . 70max_script_time configuration option . . . . 121maxage attribute . . . . . . . . . . . . . .56, 57maxage parameter . . . . . . . . . . . . .39, 40maxincalltime configuration option . . . . . . 120max-page-count configuration option . . . . 208max-page-size configuration option . . . . . 209maxstale attribute . . . . . . . . . . . . .56, 57maxstale parameter . . . . . . . . . . . .39, 40mcp-asr-usage-mode configuration option . . 144mcp-asr-usage-mode parameter . . . . . . . 40mcp-max-log-level configuration option . . . 144mcp-max-log-level parameter . . . . . . . . 40mcp-sendrecv-enabled configuration option . 145mcp-sendrecv-enabled parameter . . . . . . 40media

path . . . . . . . . . . . . . . . . . . . 43, 44services . . . . . . . . . . . . . . . . . . . 43

Media Control Platformapplication modules . . . . . . . . . . . . 255CDR attributes. . . . . . . . . . . . . . . . 67CDRs . . . . . . . . . . . . . . . . . . . . 42components . . . . . . . . . . . . . . . . . 38conference configuration options . . . . . 103configuring. . . . . . . . . . . . . . . . . 158customizing SIP responses . . . . . . . . 117

data in SIP headers . . . . . . . . . . . . 295default SIP transports . . . . . . . . . . . . 93enabling ASR and TTS . . . . . . . . . . 159functions . . . . . . . . . . . . . . . . . . 22grammars . . . . . . . . . . . . . . . . . . 44implied grammars . . . . . . . . . . . . . . 45logs . . . . . . . . . . . . . . . . . . . . . 42Media Server . . . . . . . . . . . . . . . . 38media services . . . . . . . . . . . . . . . 42metrics . . . . . . . . . . . . . . . . . . . 42module IDs . . . . . . . . . . . . . . . . 255MRCP Client . . . . . . . . . . . . . . . . 38NGI . . . . . . . . . . . . . . . . . . . . . 38Reporting Client. . . . . . . . . . . . . . . 63role and functioning . . . . . . . . . . . 38, 44role in call flow . . . . . . . . . . . . . 26, 27session timers. . . . . . . . . . . . . . . 119SNMP . . . . . . . . . . . . . . . . . . . . 70specifier IDs. . . . . . . . . . . . . . . . 256SRTP . . . . . . . . . . . . . . . . . . . . 95upstream metrics . . . . . . . . . . . . . . 65See also Media Server, NGI

media controllerspecifier IDs. . . . . . . . . . . . . . . . 269

media controller events . . . . . . . . . . . 269media gateway

role in call flow . . . . . . . . . . . . . 26, 27media negotiation

media-less dialog . . . . . . . . . . . . . . 41offer . . . . . . . . . . . . . . . . . . . . . 40SDP offer/answer . . . . . . . . . . . . . . 43

Media Serverdescribed . . . . . . . . . . . . . . . . . . 38features . . . . . . . . . . . . . . . . . . . 43services . . . . . . . . . . . . . . . . . . . 42

mediacontroller configuration section. . . . . . . . . . . 103, 186, 187

mediactrller configuration section . . . 103, 186MEDIAREDIRECT transfer . . . . . . . . 47, 49memory

Fetching Module shared . . . . . . 38, 51, 55message brokers . . . . . . . . . . . . . . 109message_format configuration option . . . . 113messaging configuration section. . . . . . . 207metadata, for configuration options . . . . . .81method

attribute (VoiceXML) . . . . . . . . . . . . 47parameter . . . . . . . . . . . . . . . . 39, 52transfer . . . . . . . . . . . . . . . . . . . 46

methodsfetch. . . . . . . . . . . . . . . . . . . . . 52get. . . . . . . . . . . . . . . . . . . . . . 52post . . . . . . . . . . . . . . . . . . . . . 52

metricsand CDRs . . . . . . . . . . . . . . . . . . 64batching . . . . . . . . . . . . . . . . . . . 68

Index

User’s Guide 333

default filters . . . . . . . . . . . . . . . . 110defined. . . . . . . . . . . . . . . . . . . . 63delivery . . . . . . . . . . . . . . . . . . . 65described . . . . . . . . . . . . . . . . . . 64examples . . . . . . . . . . . . . . . . . . 64filter . . . . . . . . . . . . . . . .65, 105, 107filter (IVR Profile) . . . . . . . . . . . . . 143generated . . . . . . . . . . . . . . . . 42, 53transfer method/type implications . . . . . . 49upstream, defined . . . . . . . . . . . . . . 65upstream, not supported . . . . . . . . . 104VAR . . . . . . . . . . . . . . . . . . 65, 300VAR prefix . . . . . . . . . . . . . . . . . . 65

metricsconfig.<Sink Name> configuration option . . . . . . . . . . . . . . . . 107

metricsfilter configuration option . . . . . . . 143MFSINK

described . . . . . . . . . . . . . . . . . . 65MIBs . . . . . . . . . . . . . . . . . .23, 66, 70min_se configuration option . . . . . . . . . 119mixed audio/video

file formats. . . . . . . . . . . . . . . . . 291file formats, record. . . . . . . . . . . . . 294services . . . . . . . . . . . . . . . . . . . 43

modifyingresource groups . . . . . . . . . . . . . . 129service requests . . . . . . . . . . . . . . . 32

module IDsdefined. . . . . . . . . . . . . . . . . . . . 64listed . . . . . . . . . . . . . . . . . . . . 255

monitor configuration section. . . . 92, 125, 126monitoring

Genesys Administrator . . . . . . . . . . 219reporting . . . . . . . . . . . . . . . .219, 227resource health . . . . . . . . . . . . . . . 33

monitor-method configuration option . . . . . 132monospace font . . . . . . . . . . . . . . . 14mpc configuration section . . . . . . . . . . 165MRCP

client/server dialog . . . . . . . . . . . . . 44services load-balanced . . . . . . . . . . . 34

MRCP Clientdescribed . . . . . . . . . . . . . . . . . . 38role . . . . . . . . . . . . . . . . . . . . . 34role and functioning . . . . . . . . . . . . . 44

MRCP serverassigning . . . . . . . . . . . . . . . . . 162role in call flow. . . . . . . . . . . . . . . . 27

MRCPv1application templates . . . . . . . . . . . 160control sessions . . . . . . . . . . . . . . . 44speech servers in GVP . . . . . . . . . . 160

MRCPv2application templates . . . . . . . . . . . 160control sessions . . . . . . . . . . . . . . . 44speech servers in GVP . . . . . . . . . . 160

mtinternal configuration section . . . . . . . 162multiple codecs . . . . . . . . . . . . . . . .44

Nnamelist parameter . . . . . . . . . . . . . .52navigation, using DTMF keys . . . . . . . . .43NETANN . . . . . . . . . . . . . . . 38, 50, 162Network Load Balancing. . . . . . . . . . . .37new call (CDR call type) . . . . . . . . . . . .67Next Generation Interpreter

See NGINGI

and VoiceXML debugger . . . . . . . . . . 50configuration options . . . . . . . . . . . 163described . . . . . . . . . . . . . . . . . . 38module IDs . . . . . . . . . . . . . . . . 265role and functioning . . . . . . . . . . . 40, 41role in call flow . . . . . . . . . . . . . . . 26shared memory . . . . . . . . . . . . . . . 55specifier IDs. . . . . . . . . . . . . . . . 265

NLB . . . . . . . . . . . . . . . . . . . 37, 134NLB.bat file . . . . . . . . . . . . . . . . . 136no_cache_url_substr configuration option . . 199noresource-response-code

configuration option . 116, 130, 132, 287

Ooffer/answer mechanism. . . . . . . . . . . .43one-leg transfers. . . . . . . . . . . . . . . .46openssl . . . . . . . . . . . . . . . . . . . .96OpenSSL Toolkit . . . . . . . . . . . . . . . .96Operational Reporting

See OR reportingOPTIONS messages, SIP . . . . . . . . . . 132OPTIONS.header.Accept configuration

option . . . . . . . . . . . . . . . . 117OPTIONS.header.Accept-Encoding

configuration option . . . . . . . . . 117OPTIONS.header.Accept-Language

configuration option . . . . . . . . . 117OPTIONS.header.Allow configuration

option . . . . . . . . . . . . . . . . 117options_response_contenttype configuration

option . . . . . . . . . . . . . . . . 116options_response_msg_body configuration

option . . . . . . . . . . . . . . . . 116OR interface

role and functioning . . . . . . . . . . . . . 68OR reporting

call arrivals . . . . . . . . . . . . . . . . . 68call peaks . . . . . . . . . . . . . . . . . . 68interface. . . . . . . . . . . . . . . . . . . 62summarization . . . . . . . . . . . . . . . 69

Index


OR statisticsdelivery to Reporting Server. . . . . . . . . 68

ors.reportinginterval configuration option . . 107out.<SIP request>.headers configuration

option . . . . . . . . . . . . . . . . 172out.<SIP request>.params configuration

option . . . . . . . . . . . . . . . . 172outbound

proxy. . . . . . . . . . . . . . . . . . . . . 53route set . . . . . . . . . . . . . . . . . . . 87

outbound (CDR call type) . . . . . . . . . . 66outbound-call-allowed configuration option . 145outbound-call-forbidden-respcode

configuration option . . . . . . . . . 117outbound-call-forbidden-set-alarm

configuration option . . . . . . . . . 117outboundproxy hint . . . . . . . . . . . . . . 53outbound-usage-limit configuration option . . 145outbound-usage-limit-exceeded-set-alarm

configuration option . . . . . . . . . 117outcalluseoriggw configuration option . . . . 173outgoing connections. . . . . . . . . . . . . 53output control, conference . . . . . . . . . . 43

Pparameters

ccxml . . . . . . . . . . . . . . . . . . . . 51confmaxsize . . . . . . . . . . . . . . . 34, 35confreserve . . . . . . . . . . . . . . . 34, 35enctype . . . . . . . . . . . . . . . . . . . 52for service and policies . . . . . . . . . . . 32gvp.alternatevoicexml . . . . . . . . . . . . 39gvp.config . . . . . . . . . . . . . . . 39, 294gvp.policy . . . . . . . . . . . . . . . . . . 40gvp.rm.tenant-id . . . . . . . . . . . . . . . 73gvp-tenant-id . . . . . . . . . . . . . . . . 73HTTP/HTTPS fetch . . . . . . . . . . . . . 52lr . . . . . . . . . . . . . . . . . . . . . . . 32maxage . . . . . . . . . . . . . . . . . 39, 40maxstale . . . . . . . . . . . . . . . . . 39, 40mcp-asr-usage-mode . . . . . . . . . . . . 40mcp-max-log-level . . . . . . . . . . . . . . 40mcp-sendrecv-enabled . . . . . . . . . . . 40method . . . . . . . . . . . . . . . . . . . 39namelist . . . . . . . . . . . . . . . . . . . 52postbody. . . . . . . . . . . . . . . . . . . 39Request-URI. . . . . . . . . . . . . 30, 31, 34Request-URI, for conference . . . . . . . . 35service, in Request-URI . . . . . . . . . . . 39timeout. . . . . . . . . . . . . . . . . . 39, 40trunkport . . . . . . . . . . . . . . . . . . . 30vendor-specific, for TTS . . . . . . . . . . 161voicexml . . . . . . . . . . . . . . . . . . . 39See also configuration options

pathsdefault SSL private key and certificate . . . 97log files . . . . . . . . . . . . . . . . . . . 64persistent queue for CDRs . . . . . . . . 108

pcma . . . . . . . . . . . . . . . . . . . . . .44pcmu . . . . . . . . . . . . . . . . . . . . . .44performance

and reporting summarization . . . . . . . . 69and SSL. . . . . . . . . . . . . . . . . . . 71

persistence configuration section . . . . . . 207persistent queue . . . . . . . . . . . . . . . .68

path . . . . . . . . . . . . . . . . . . . . 108platform debugging . . . . . . . . . . . . . .50platform.save_<file type>_files configuration

option . . . . . . . . . . . . . . . . 187play

audio file formats . . . . . . . . . . . . . 289audio/video file formats . . . . . . . . . . 291video file formats . . . . . . . . . . . . . 291

policiescaching . . . . . . . . . . . . . . . . . . . 56database retention . . . . . . . . . . 140, 206enforcing . . . . . . . . . . . . . . . . . . 32IVR Profile . . . . . . . . . . . . . . 28, 143

port-capacity configuration option . . . . . . 130port-usage-type configuration option . . 34, 133post method . . . . . . . . . . . . . . . . . .52postbody parameter . . . . . . . . . . . . . .39prefixes

dialing, hookflash transfer. . . . . . . . . 170userdata variables . . . . . . . . . . . . 178VAR metrics . . . . . . . . . . . . . . . . . 65

private keycreating . . . . . . . . . . . . . . . . . 96, 97default path . . . . . . . . . . . . . . . . . 97SSL . . . . . . . . . . . . . . . . . . . . . 95

promptsaudio file formats . . . . . . . . . . . . . 289audio/video file formats . . . . . . . . . . 291video file formats . . . . . . . . . . . . . 291

properties, device profile. . . . . . . . . . . 189protocols

preferred SIP . . . . . . . . . . . . . . . . 86provisioning

ASR resources . . . . . . . . . . . . . . 160device profiles. . . . . . . . . . . 54, 188, 194GVP . . . . . . . . . . . . . . . . . . . . . 82IVR Profiles . . . . . . . . . . . . . . . . 139resources . . . . . . . . . . . . . . . . . 126TTS resources . . . . . . . . . . . . . . 160

proxy configuration section . . . . . . . 92, 125purging

cache objects . . . . . . . . . . . . . . . . 60reporting data . . . . . . . . . . . . . . . . 69

pwproxy . . . . . . . . . . . . . . . . . . . .55

Index

User’s Guide 335

Qquartz.rs.calltimeout configuration option . . 210quartz.rs.dbMaintenancePeriod configuration

option . . . . . . . . . . . . . . . . 210queue, persistent . . . . . . . . . . . . . . . 68

Rrc.batch_size configuration option . . . . . . 107rc.cdr.batch_size configuration option . . . . 108rc.cdr.local_queue_path configuration

option . . . . . . . . . . . . . . . . 108rc.cdr.msg_broker_uri configuration option . 108rc.local_queue_path configuration option . . 109rc.msg_broker_uri configuration option . . . 109rc.ors.msg_broker_uri configuration option . 109rc.ors_local_queue_path configuration

option . . . . . . . . . . . . . . . . 109Real-time Transport Protocol

See RTPReason header . . . . . . . . . . . . . . 42, 286record

audio file formats . . . . . . . . . . . . . 292audio/video file formats . . . . . . . . . . 294video file formats . . . . . . . . . . . . . 293

recording . . . . . . . . . . . . . . . . . . . 43DTMF input . . . . . . . . . . . . . . . . . 43

REFER transfer . . . . . . . . . . . . . .46, 48REFER with replaces transfer . . . . . . .46, 49REFERJOIN transfer . . . . . . . . . . . .46, 49referxferhold configuration option . . . . . . 173refreshing cache objects . . . . . . . . . . . 60refresh-pattern rules . . . . . . . . . . . . . 201Refresh-Rate model . . . . . . . . . . . . . 59registrar configuration section . . . . . . 92, 125registration configuration option . . . . . . . 174related resources. . . . . . . . . . . . . . . 15RelaxNG schemas . . . . . . . . . . . . . . 69report filters. . . . . . . . . . . . . . . . . . 224reporting

active call list . . . . . . . . . . . . . . . 228architecture . . . . . . . . . . . . . . . . . 62call completion summary . . . . . . . . . 247categories . . . . . . . . . . . . . . . . . . 69configuration options . . . . . . . . . . . 104connecting to Genesys Administrator . . . 213controlling access . . . . . . . . . . . 69, 211credentials . . . . . . . . . . . . . . . . . . 69database maintenance . . . . . . . . . . . 69database retention periods . . . . . . . . 206database retention policies . . . . . .140, 206enabling HTTPS. . . . . . . . . . . . . . 215enabling in GVP . . . . . . . . . . . . . . . 82exponential back-off . . . . . . . . . . . . . 68granularity . . . . . . . . . . . . . . . . . 204

historical call browser . . . . . . . . . . . 238historical call summary . . . . . . . . . . 235historical peaks . . . . . . . . . . . . . . 236historical reports . . . . . . . . . . . . . 233HTTP Basic Authorization . . . . . . . . 211ivr action usage . . . . . . . . . . . . . . 248last ivr action used . . . . . . . . . . . . 249overview . . . . . . . . . . . . . . . 60, 219performance considerations . . . . . . . . 69persistent queue . . . . . . . . . . . . . . 68real-time reports. . . . . . . . . . . . . . 227report categories . . . . . . . . . . . . . . 69report filters . . . . . . . . . . . . . . . . 224retention. . . . . . . . . . . . . . . . . . 234running a report . . . . . . . 219, 227, 233, 243statistics . . . . . . . . . . . . . . . . . . . 66summarization process . . . . . . . . . . . 69upstream, defined . . . . . . . . . . . . . . 65VAR call browser . . . . . . . . . . . . . 244voice application reports . . . . . . . . . 243XML schemas. . . . . . . . . . . . . . . . 69

Reporting Clientpersistent queue . . . . . . . . . . . . . . 68role and functioning . . . . . . . . . . . 63, 68

reporting configuration section. . . . . 207, 210Reporting Server

configuring . . . . . . . . . . . . . . . . 203connecting to Genesys Administrator . . . 213functions . . . . . . . . . . . . . . . . . . 23message brokers . . . . . . . . . . . . . 109role and functioning . . . . . . . . . 63, 66, 68services . . . . . . . . . . . . . . . . . . . 68web server and HTTPS . . . . . . . . . . . 95

Reporting Web Servicesaccess control. . . . . . . . . . . . . . . . 69default URL . . . . . . . . . . . . . . . . . 69described . . . . . . . . . . . . . . . . . . 69

request failuresResource Manager handling . . . . . . . . 36

Request-URIformats . . . . . . . . . . . . . . . . . . . 39parameters . . . . . . . . . . . 30, 31, 34, 40parameters, encoded . . . . . . . . . . 39, 40parameters, for conference . . . . . . . . . 35service parameters . . . . . . . . . . . . . 39

resource groupsadding resources . . . . . . . . . . . . . 129configuring . . . . . . . . . . . . . . 126, 127deleting . . . . . . . . . . . . . . . . . . 129load-balancing . . . . . . . . . . . . . . . 34management wizard . . . . . . . . . . . 128modifying properties . . . . . . . . . . . 129service types . . . . . . . . . . . . . . . . 33

Resource Managerand upstream metrics . . . . . . . . . . . 104as SIP registrar . . . . . . . . . . . . . . . 33

Index


cluster mode. . . . . . . . . . . . . . . . 136configuring. . . . . . . . . . . . . . . . . 124customizing SIP responses . . . . . . . . 116default SIP transports . . . . . . . . . . . . 93failed request handling . . . . . . . . . . . 36functions . . . . . . . . . . . . . . . . . . . 21High Availability . . . . . . . . . . . . 37, 134load-balancing resources . . . . . . . . . . 34logical resource groups . . . . . . . . . . . 33managing resources. . . . . . . . . . 32, 127modifying service requests . . . . . . . . . 32module IDs . . . . . . . . . . . . . . . . 271policy enforcement . . . . . . . . . . . . . 32Reporting Client . . . . . . . . . . . . . . . 63resource selection . . . . . . . . . . . . . . 33role and functioning . . . . . . . . . . . . . 27role in call flow. . . . . . . . . . . . . . . . 26selecting resources for conference . . . . . 34service selection. . . . . . . . . . . . . 29, 31session management . . . . . . . . . 27, 118session timers . . . . . . . . . . . . . 118, 119SNMP . . . . . . . . . . . . . . . . . . . . 70specifier IDs . . . . . . . . . . . . . . . . 271

resource-confmaxcount configuration option . . . . . . . . . . . . . . . . 134

resource-confmaxsize configuration option . 133resource-no-match-respcode configuration

option . . . . . . . . . . . . . . . . 116resources

assigning to groups . . . . . . . . . .129, 133gateway . . . . . . . . . . . . . . . . . . 129health monitoring . . . . . . . . . . . . . . 33High Availability . . . . . . . . . . . . . . . 37management wizard . . . . . . . . . . . . 128managing . . . . . . . . . . . . . . . 32, 127provisioning . . . . . . . . . . . . . . . . 126provisioning ASR and TTS . . . . . . . . 160selecting . . . . . . . . . . . . . . . . . . . 33selecting for conference . . . . . . . . . . . 34

resource-unavailable-respcode configuration option . . . . . . . . . . . . . . . . 116

rm configuration section . . . . . . . . . . . 125roles

<log> tag interface. . . . . . . . . . . . . . 63Call Control Platform . . . . . . . . . . . . 51CCXMLI . . . . . . . . . . . . . . . . . . . 52CDR Service. . . . . . . . . . . . . . . 62, 66DATAC. . . . . . . . . . . . . . . . . . . . 65EMS Logging interface . . . . . . . . . . . 62log sinks . . . . . . . . . . . . . . . . . . . 65Media Control Platform . . . . . . . . . . . 38MRCP Client. . . . . . . . . . . . . . . . . 44NGI . . . . . . . . . . . . . . . . . . . 40, 41OR interface . . . . . . . . . . . . . . . . . 68OR service. . . . . . . . . . . . . . . . . . 62Reporting Client . . . . . . . . . . . . . 63, 68

Reporting Server . . . . . . . . . . 63, 66, 68Reporting Web Services . . . . . . . . . . 69Resource Manager . . . . . . . . . . . . . 27

root page, caching . . . . . . . . . . . . . 52, 58round robin load balancing. . . . . . . . . . 131round robin load-balancing . . . . . . . . . .34Route header . . . . . . . . . . . . . . . . .32route set

configuring . . . . . . . . . . . . . . . . . 87route.default.<protocol> configuration option .88route.dest.<n> configuration option . . . . 87, 89routeset configuration option. . . . . . . . 87, 90routing

configuring . . . . . . . . . . . . . . . 86, 87rs.db.retention.<type of data>.default

configuration option . . . . . . . . . 209rs.httpport configuration option . . . . . . . 215rs.password configuration option . . . . . . 215rs.query.limit.<granularity> configuration

options . . . . . . . . . . . . . . . 205rs.query.limit.<time period> configuration

option . . . . . . . . . . . . . . . . 210rs.username configuration option . . . . . . 216RTP media path . . . . . . . . . 27, 43, 44, 162rtp.timeout configuration option . . . . . . . 120RTSP . . . . . . . . . . . . . . . . . . . . .44rule-<n> configuration option . . . . . . . . 149rules, dialing . . . . . . . . . . . . . . . . . 148running a report . . . . . . . . . . . . . . . 219

Ssafe ports, Squid . . . . . . . . . . . . . . . 202schedule configuration section. . . . . 207, 210screen captures . . . . . . . . . . . . . . . .15SDP

answer . . . . . . . . . . . . . . . . . . . 41codecs offered . . . . . . . . . . . . . . . 44offer . . . . . . . . . . . . . . . . . . . . . 40offer/answer . . . . . . . . . . . . . . . . . 43

sectionsSee configuration sections

secure communicationsenabling . . . . . . . . . . . . . . . . . . . 94implementing . . . . . . . . . . . . . . . . 71supported . . . . . . . . . . . . . . . . . . 70

secure HTTPSee HTTPS

secure RTPSee SRTP

secure SIPSee SIPS

Secure Socket LayerSee SSL

securerouteset configuration option . . . . 87, 91

Index

User’s Guide 337

securitySee secure communications

segment configuration option . . . . . . . . 114selecting

ccxml service . . . . . . . . . . . . . . . . 31conference resources . . . . . . . . . . . . 34conference service . . . . . . . . . . . . . 31device profiles . . . . . . . . . . . . . . . . 54gatewayservice . . . . . . . . . . . . . . . 31resources . . . . . . . . . . . . . . . . . . 33services . . . . . . . . . . . . . . . . . 29, 31voicexml service . . . . . . . . . . . . . . . 31

self-signed SSL certificate . . . . . . . . . . 97sendalert configuration option . . . . . 117, 175service requests

load-balancing . . . . . . . . . . . . . . . . 34modifying . . . . . . . . . . . . . . . . . . 32specified in Request-URI . . . . . . . . . . 31

servicesaudio, video, mixed . . . . . . . . . . . . . 43bridging . . . . . . . . . . . . . . . . . . . 52capabilities. . . . . . . . . . . . . . . . . 145ccxml . . . . . . . . . . . . . . . . . . . . 31conference. . . . . . . . . . . . . . . . 31, 52conference, configuring . . . . . . . . . . 102configuring types . . . . . . . . . . .133, 142external SIP . . . . . . . . . . . . . . . . . 32gateway . . . . . . . . . . . . . . . . . . . 31Media Server . . . . . . . . . . . . . . . . 42prerequisites. . . . . . . . . . . . . . . . 150Reporting Server . . . . . . . . . . . . . . 68resource groups . . . . . . . . . . . . . . . 33selecting . . . . . . . . . . . . . . . . . 29, 31transcoding . . . . . . . . . . . . . . . . . 52types. . . . . . . . . . . . . . . . . .133, 142voicexml . . . . . . . . . . . . . . . . . . . 31

service-type configuration option. . . . 103, 142service-types configuration option . . . . . . 133session

expiry timers . . . . . . . . . . . . . . . . . 28identifiers . . . . . . . . . . . . . . . . . . 72inactivity timers . . . . . . . . . . . . . . . 28management . . . . . . . . . . . . . . . . 27timer configuration options . . . . . . . . 118timers . . . . . . . . . . . . . . 117, 142, 155

session configuration section . . . . . . . . 186Session Description Protocol

See SDPsession variables

session.com.genesyslab.userdata . . . . . 42Session ID . . . . . . . . . . . . . . . . . . 72session.com.genesyslab.userdata session

variable . . . . . . . . . . . . . . . 42sessionexpires configuration option . . . . . 119Session-Expires header . . . . . . 29, 118, 119sessionparams configuration option . . . . . 95

sessionparamsoffer configuration option . . .95sessmgr configuration section . . . . . . 95, 162Settings tab . . . . . . . . . . . . . . . . . .78

changing the display . . . . . . . . . . . . 79Display Filter . . . . . . . . . . . . . . . . 80

severitylogs . . . . . . . . . . . . . . . . . . . . . 63

shadow variables . . . . . . . . . . . . . . .42Simple Network Management Protocol

See SNMPsinks, log

See log sinksSIP

BRIDGE transfer method . . . . . . . . 46, 48configuring communications . . . . . . . . 86default transports . . . . . . . . . . . . . . 93device capabilities. . . . . . . . . . . . . . 53external service . . . . . . . . . . . . . . 133HKF transfer method . . . . . . . . . . 46, 47INFO messages. . . . . . . . . . . . 41, 169MEDIAREDIRECT transfer method. . . 47, 49OPTIONS messages . . . . . . . . . . . 132REFER transfer method . . . . . . . . 46, 48REFERJOIN transfer method . . . . . . 46, 49stack . . . . . . . . . . . . . . . . . . . . 87transports . . . . . . . . . . . . . . . . . . 86

sip configuration section . . 92, 95, 163, 167, 186SIP headers

configuring DNIS source . . . . . . . . . . 30custom . . . . . . . . . . . . . . . . . 40, 42Reason . . . . . . . . . . . . . . . . . . . 42Route . . . . . . . . . . . . . . . . . . . . 32Session-Expires. . . . . . . . . . . . . . . 29used by Media Control Platform . . . . . 295X-Genesys-CallUUID . . . . . . . . . . . . 72X-Genesys-GVP-Session-ID . . . . . . 28, 72X-Genesys-RM-Application-dbid . . . . 28, 73

SIP parametersfor service and policies . . . . . . . . . . . 32gvp.rm.datanode . . . . . . . . . . . . . . 28gvp.rm.tenant-id. . . . . . . . . . . . . . . 28

SIP responses . . . . . . . . . . . . . . . . 298100 Trying. . . . . . . . . . . . . . . . . 279180 Ringing . . . . . . . . . . . . . . . . 280183 Session Progress . . . . . . . . 43, 280202 Accepted . . . . . . . . . . . . . . . 2812xx . . . . . . . . . . . . . . . . . . . . . 36302 Moved Temporarily . . . . . . . . . . 2813xx . . . . . . . . . . . . . . . . . . . . 281400 Bad Request . . . . . . . . . . . . . 281403 Forbidden . . . . . . . . . . . . . . 282404 Not Found . . . . . . . . . . .30, 32, 282405 Method Not Allowed . . . . . . . . . 282408 Request Timeout . . . . . . . . . 36, 283420 Bad Extension . . . . . . . . . . . . 283423 Interval Too Brief . . . . . . . . . . . 283

Index


480 Temporarily Unavailable . . . . . . . 283487 Request Terminated . . . . . . . . . 284488 Not Acceptable Here . . . . . . . . . 2844xx or 5xx . . . . . . . . . . . . . . . . . . 36500 Server Internal Error . . . . . . . . . 285503 Service Unavailable. . . . . . . . . . 2856xx. . . . . . . . . . . . . . . . . . . . . . 36customizing . . . . . . . . . . . . . . 116, 117gateway failures . . . . . . . . . . . . . . 130in GVP. . . . . . . . . . . . . . . . . . . 279

SIP Server, role in call flow. . . . . . . . . . 26sip.allowedunknownheaders configuration

option . . . . . . . . . . . . . . . . 186sip.min_se configuration option . . . . . . . 119sip.proxy.optionsinterval configuration option 126sip.proxy.unavailoptionsinterval configuration

option . . . . . . . . . . . . . . . . 126sip.registrar.maxexpirytime configuration

option . . . . . . . . . . . . . . . . 120sip.registrar.minexpirytime configuration

option . . . . . . . . . . . . . . . . 120sip.route.default.<protocol> configuration

option . . . . . . . . . . . . . . . . 88sip.route.dest.<n> configuration option . . . 89sip.route.dests configuration option . . . . . 90sip.send_progressing configuration option . . 117sip.sessionexpires configuration option . . . 119sip.sessiontimer configuration

option . . . . . . . . . . . 118, 142, 155sip.timer.ci_proceeding configuration

option . . . . . . . . . . . . . . . . 120sip.transport.<x> configuration option . . 92, 136sip.transport.<x> configuration options. . . . 86sip-header-for-dnis configuration option . 30, 126sipinfoallowedcontenttypes configuration

option . . . . . . . . . . . . . . . . 175sipinfodtmf configuration option . . . . . . . 41sipproxy configuration option. . . . . . . . . 188SIPS

enabling . . . . . . . . . . . . . . . . . . . 94supported . . . . . . . . . . . . . . . . 70, 94

SNMPMaster Agent . . . . . . . . . . . . . . . . 70support . . . . . . . . . . . . . . . . . . . 70

SNMP integration sink . . . . . . . . . . . . 66specifications and standards . . . . . . . . . 307specifiers

defined. . . . . . . . . . . . . . . . . . . . 64IDs . . . . . . . . . . . . . . . . . . . . . 255

square brackets . . . . . . . . . . . . . . . 15Squid

access log files . . . . . . . . . . . . 60, 317cache management . . . . . . . . . . . . . 60caching . . . . . . . . . . . . . . . . . . . 56caching algorithm . . . . . . . . . . . . . 316caching behavior . . . . . . . . . . . . . . 57

caching model . . . . . . . . . . . . . . . 59caching policies . . . . . . . . . . . . . . . 56clearing the cache . . . . . . . . . . . . . 60configuration file . . . . . . . . . . . . . . 59configuring . . . . . . . . . . . . . . . . 200expiry time algorithm . . . . . . . . . . . 316functions . . . . . . . . . . . . . . . . . . 22logs . . . . . . . . . . . . . . . . . . . . . 60purging cache objects. . . . . . . . . . . . 60refreshing cache objects . . . . . . . . . . 60refresh-pattern rules . . . . . . . . . . . 201Refresh-Rate model . . . . . . . . . . . . 59role . . . . . . . . . . . . . . . . . . . . . 55safe ports . . . . . . . . . . . . . . . . . 202SSL ports . . . . . . . . . . . . . . . . . 202

SRTPenabling . . . . . . . . . . . . . . . . . 94, 95encryption keys . . . . . . . . . . . . . . . 71supported . . . . . . . . . . . . . . . . . . 71

srtp.cryptomethods configuration option. . . .95srtp.mode configuration option. . . . . . 95, 166SSL

certificate . . . . . . . . . . . . . . . . . . 95certificate, creating . . . . . . . . . . . . . 96configuring Tomcat . . . . . . . . . . . . 101default certificate path. . . . . . . . . . . . 97default private key path . . . . . . . . . . . 97Fetching Module configuration options . . 199performance considerations . . . . . . . . 71ports, Squid . . . . . . . . . . . . . . . . 202private key . . . . . . . . . . . . . . . . . 95private key, creating. . . . . . . . . . . 96, 97self-signed certificate, creating . . . . . . . 97support . . . . . . . . . . . . . . . . . . . 52supported . . . . . . . . . . . . . . . . . . 70

ssl_* configuration options. . . . . . . . . . 199ssl_cipher_list configuration option . . . . . 101ssl_key configuration option . . . . . . . . . 100ssl_key_passwd configuration option . . . . 100stack configuration section. . . . . . . . . . 163standard configuration option . . . . . . . . 112standards, supported . . . . . . . . . . . . 307starting

GVP . . . . . . . . . . . . . . . . . . . . . 55statistics

call arrivals . . . . . . . . . . . . . . . . . 68call peaks . . . . . . . . . . . . . . . . . . 68summarization . . . . . . . . . . . . . . . 69VAR . . . . . . . . . . . . . . . . . . . . . 69

statistics, calculated . . . . . . . . . . . . . .66subagent, AgentX (for SNMP) . . . . . . . . .70summarization

statistics . . . . . . . . . . . . . . . . . . . 69supervised transfer type . . . . . . . . . . 45, 47support

HTTPS . . . . . . . . . . . . . . . . . . . 70

Index

User’s Guide 339

secure communications . . . . . . . . . . . 70SIPS . . . . . . . . . . . . . . . . . . . 70, 94specifications and standards . . . . . . . 307SRTP . . . . . . . . . . . . . . . . . . . . 71SSL . . . . . . . . . . . . . . . . . . . . . 70TLS . . . . . . . . . . . . . . . . . . . . . 70upstream metrics . . . . . . . . . . . . . 104

supported transfer methods . . . . . . . . . 175suspend-mode-respcode configuration

option . . . . . . . . . . . . . . . . 116

Ttask summary

configuring GVP . . . . . . . . . . . . . . . 82provisioning GVP . . . . . . . . . . . . . . 82

telephone-event . . . . . . . . . . . . . . . 44tenant

assigning default IVR Profile . . . . .140, 154IVR Profile settings . . . . . . . . . . . . 153

tenant, Environmentdefault IVR Profile . . . . . . . . . . . . . 140session timers . . . . . . . . . . . . . . . 118

Text-to-SpeechSee TTS

tfci . . . . . . . . . . . . . . . . . . . . . . 44Third Party Squid

See Squidtime-format configuration option . . . . . . . 114timeout parameter . . . . . . . . . . . . .39, 40timeouts . . . . . . . . . . . . . . . . 117, 120timers. . . . . . . . . . . . . . . . . . . . . 120

inactivity . . . . . . . . . . . . . . . . . . 118session . . . . . . . . . . . . . 117, 142, 155session expiry . . . . . . . . . . . . . . . 118

TLSand CCXML applications . . . . . . . . . . 94SIP transport . . . . . . . . . . . . . . . . 94supported . . . . . . . . . . . . . . . . . . 70

TLSv1 . . . . . . . . . . . . . . . . . . . . 101Tomcat . . . . . . . . . . . . . . . . . . 95, 101trace configuration option . . . . . . . . . . 112trace_flag configuration option . . . . . . . . 110transaction configuration section . . . . . . . 207transcoding

implicit . . . . . . . . . . . . . . . . . . . . 52services . . . . . . . . . . . . . . . . . . . 52

transfer.allowed configuration option . . . . . 178transfer-allowed configuration option. . . . . 147transfer-forbidden-respcode configuration

option . . . . . . . . . . . . . . . . 117transfer-forbidden-set-alarm configuration

option . . . . . . . . . . . . . . . . 117transfermethods configuration option . . . . 175transfers

blind . . . . . . . . . . . . . . . . . . . 45, 47

bridge . . . . . . . . . . . . . . . . . . 45, 48BRIDGE method . . . . . . . . . . . . 46, 48consultation . . . . . . . . . . . . . . . 45, 47default methods . . . . . . . . . . . . . . 167dialog-initiated . . . . . . . . . . . . 52, 178HKF method . . . . . . . . . . . . . . 46, 47implications for metrics . . . . . . . . . . . 49join-style . . . . . . . . . . . . . . . . . . 46MEDIAREDIRECT method . . . . . . . 47, 49methods. . . . . . . . . . . . . . . . . . . 46one-leg . . . . . . . . . . . . . . . . . . . 46REFER method . . . . . . . . . . . . . 46, 48REFERJOIN method . . . . . . . . . . 46, 49supervised . . . . . . . . . . . . . . . 45, 47supported methods . . . . . . . . . . . . 175two-leg . . . . . . . . . . . . . . . . . . . 46types . . . . . . . . . . . . . . . . . . . . 45whisper . . . . . . . . . . . . . . . . . . . 45

transport.<x> configuration option . . . . . . .92transports

default . . . . . . . . . . . . . . . . . . . . 93transports, SIP

for Resource Manager . . . . . . . . . . . 86preferred protocol . . . . . . . . . . . . . . 86

trapsin GVP . . . . . . . . . . . . . . . . . . . 70TRAPSINK . . . . . . . . . . . . . . . . . 66

TRAPSINKdescribed . . . . . . . . . . . . . . . . . . 66

trunkport parameter . . . . . . . . . . . . . .30TTS

configuration options . . . . . . . . . . . 179enabling . . . . . . . . . . . . . . . . . . 159in GVP . . . . . . . . . . . . . . . . . . . 44provisioning resources . . . . . . . . . . 160vendor-specific parameters . . . . . . . . 161

two-leg transfers . . . . . . . . . . . . . . . .46typographical styles . . . . . . . . . . . . . .14tzoffset configuration option . . . . . . . . . 216

UULAW . . . . . . . . . . . . . . . . . . . . 166UNIVERSALS properties . . . . . . . . . . .44unknown (CDR call type) . . . . . . . . . . .67upstream metrics

defined . . . . . . . . . . . . . . . . . . . 65not supported . . . . . . . . . . . . . . . 104

upstream reporting, defined . . . . . . . . . .65URI

default CCXML application . . . . . . . . 186URL

for Genesys Administrator . . . . . . . . . 24for Reporting Web Services. . . . . . . . . 69initial page. . . . . . . . . . . . . . . . . 151

Index


usage-limit-exceeded-respcode configuration option . . . . . . . . . . . . . 117, 147

usage-limit-exceeded-set-alarm configuration option . . . . . . . . . . . . . 117, 147

usage-limits configuration option . . . . 147, 155user data . . . . . . . . . . . . . . . . . . . 42userdata attribute. . . . . . . . . . . . . . . 42userdata variables, prefix . . . . . . . . . . 178userdata.prefix configuration option . . . . . 178use-same-gateway configuration option . . . 148using Genesys Administrator. . . . . . . . . 78UTF-16 . . . . . . . . . . . . . . . . . . . . 41UTF-8 . . . . . . . . . . . . . . . . . . . . 40

VVAR

<log> tag interface. . . . . . . . . . . . . . 63call browser report. . . . . . . . . . . . . 244metrics. . . . . . . . . . . . . . . . . 65, 300statistics . . . . . . . . . . . . . . . . . . . 69summarization . . . . . . . . . . . . . . . . 69

variablessession.com.genesyslab.userdata . . . . . 42shadow . . . . . . . . . . . . . . . . . . . 42

VCR controls . . . . . . . . . . . . . . . . . 43vendor-specific parameters, for TTS . . . . . 161verbose configuration option . . . . . . . . . 115version numbering . . . . . . . . . . . . . . 13video

file formats, play . . . . . . . . . . . . . . 291file formats, record. . . . . . . . . . . . . 293recording . . . . . . . . . . . . . . . . . . 43

video services . . . . . . . . . . . . . . . . 43Voice Application Reporter

See VARvoice application reports . . . . . . . . . . . 243Voice Platform Solution (VPS) . . . . . . .25, 77voicexml

parameter . . . . . . . . . . . . . . . . . . 39service selected . . . . . . . . . . . . . . . 31service, configuring . . . . . . . . . . . . 133

VoiceXML applicationsalternate . . . . . . . . . . . . . . . . . . 151and HTTPS . . . . . . . . . . . . . . . . . 71and IVR Profiles . . . . . . . . . . . . . . 139debugging . . . . . . . . . . . . . . . . . . 50default . . . . . . . . . . . . . . . . . . . 140dialogs . . . . . . . . . . . . . . . . . . . . 42identifiers . . . . . . . . . . . . . . . . . . 73method attribute . . . . . . . . . . . . . . . 47provisioning . . . . . . . . . . . . . . . . 139ready to proceed . . . . . . . . . . . . 40, 41receiving events . . . . . . . . . . . . . . 169sending events . . . . . . . . . . . . . . 169shadow variables . . . . . . . . . . . . . . 42

start . . . . . . . . . . . . . . . . . . . . . 41triggering . . . . . . . . . . . . . . . . . . 77VAR <log> tags . . . . . . . . . . . . . . . 63See also attributes (XML), session variables,

shadow variablesvoicexml-dialog-allowed configuration option 148voicexml-dialog-forbidden-respcode

configuration option . . . . . . . . . 117voicexml-dialog-forbidden-set-alarm

configuration option . . . . . . . . . 117voicexml-usage-limit-exceeded-respcode

configuration option . . . . . . . . . 117voicexml-usage-limit-exceeded-set-alarm

configuration option . . . . . . . . . 117vrm configuration section . . . . . . . 163, 179vrm.client.TlsCertificateKey configuration

option . . . . . . . . . . . . . . . . .95vrm.client.TlsPassword configuration option. .95vrm.client.TlsPrivateKey configuration option .95vxmli configuration section. . . . . . . 163, 177vxmli.initial_request_maxage configuration

option . . . . . . . . . . . . . . . . .58vxmli.initial_request_maxstale configuration

option . . . . . . . . . . . . . . . . .58vxmlinvite configuration option. . . . . . . . 176

Wwarningheaders configuration option . . . . 176web server

Reporting Server, and HTTPS . . . . . . . 95Tomcat . . . . . . . . . . . . . . . . . . 101

whisper transferdefined . . . . . . . . . . . . . . . . . . . 45supported . . . . . . . . . . . . . . 47, 48, 49

Windows Network Load Balancing . . . 37, 134wizard, manage resources. . . . . . . . . . 128

Xxfer.copyheaders configuration option . . . . 176X-Genesys- prefix . . . . . . . . . . . . . . 178X-Genesys-CallUUID header . . . . . . . . .72X-Genesys-GVP-Session-ID header

defined . . . . . . . . . . . . . . . . . . . 72generated . . . . . . . . . . . . . . . . . . 28parameters . . . . . . . . . . . . . . . . . 28

X-Genesys-RM-Application-dbid header . 28, 73X-Lite (device profile) . . . . . . . . . . 54, 305XML schemas, for reports . . . . . . . . . . .69

80gvp_us

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